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H.323/SIP VoIP GSM Gateway
VIP-281GS
User’s manual
Version 1.1.0
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Copyright
Copyright (C) 2007 PLANET Technology Corp. All rights reserved.
The products and programs described in this User’s Manual are licensed products of PLANET Technology, This
User’s Manual contains proprietary information protected by copyright, and this User’s Manual and all
accompanying hardware, software, and documentation are copyrighted.
No part of this User’s Manual may be copied, photocopied, reproduced, translated, or reduced to any electronic
medium or machine-readable form by any means by electronic or mechanical. Including photocopying, recording,
or information storage and retrieval systems, for any purpose other than the purchaser's personal use, and without
the prior express written permission of PLANET Technology.
Disclaimer
PLANET Technology does not warrant that the hardware will work properly in all environments and applications,
and makes no warranty and representation, either implied or expressed, with respect to the quality, performance,
merchantability, or fitness for a particular purpose.
PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims liability for any
inaccuracies or omissions that may have occurred.
Information in this User’s Manual is subject to change without notice and does not represent a commitment on the
part of PLANET. PLANET assumes no responsibility for any inaccuracies that may be contained in this User’s
Manual. PLANET makes no commitment to update or keep current the information in this User’s Manual, and
reserves the right to make improvements to this User’s Manual and/or to the products described in this User’s
Manual, at any time without notice.
If you find information in this manual that is incorrect, misleading, or incomplete, we would appreciate your
comments and suggestions.
CE mark Warning
The is a class B device, In a domestic environment, this product may cause radio interference, in which case the
user may be required to take adequate measures.
WEEE Warning
To avoid the potential effects on the environment and human health as a result of the presence of
hazardous substances in electrical and electronic equipment, end users of electrical and electronic
equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of
WEEE as unsorted municipal waste and have to collect such WEEE separately.
Trademarks
The PLANET logo is a trademark of PLANET Technology. This documentation may refer to numerous hardware
and software products by their trade names. In most, if not all cases, their respective companies claim these
designations as trademarks or registered trademarks.
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Revision
User’s Manual for PLANET H.323/SIP VoIP GSM Gateway:
Model: VIP-281GS
Rev: 1.1 (October, 2009)
Part No. EM-VIP281GSV1.1
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TABLE OF CONTENTS
Chapter 1................................................................................................ 6
Introduction............................................................................................ 6
Overview............................................................................................................................6
Package Content...............................................................................................................7
Physical Details.................................................................................................................8
Front Panel LED Indicators & Rear Panels................................................................8
Chapter 2 Preparations & Installation................................................ 10
Physical Installation Requirement................................................................................10
WAN IP address configuration via web configuration interface..............................11
Chapter 3 Network Service Configurations....................................... 12
Configuring and monitoring your VoIP Gateway from web browser.......................12
Overview on the web interface of VoIP GSM Gateway...........................................12
Manipulation of VoIP GSM Gateway via web browser...........................................12
VIP-281GS Setup for Quick Start.................................................................................13
1. Network Setup (WAN Port Type Setup)...............................................................13
2. VoIP Basic Setup:.................................................................................................15
Chapter 4 GSM Setup.......................................................................... 17
GSM Setup......................................................................................................................17
GSM Parameter ........................................................................................................18
PSTN Dialplan..........................................................................................................19
GSM Dialplan...........................................................................................................20
SMS Setup................................................................................................................20
Terminate Black List ................................................................................................21
Originate Black List .................................................................................................22
Chapter 5 Advance Setup................................................................... 23
Network Setup ................................................................................................................23
Dynamic DNS ..........................................................................................................23
Netwrok Management ..............................................................................................24
VoIP Setup .......................................................................................................................24
VoIP Basic Configuration to H.323 protocol............................................................25
Dialing Plan to H.323 protocol.................................................................................27
Advance Setting to H.323 protocol ..........................................................................30
VoIP Basic Configuration to SIP Protocol................................................................33
Dialing Plan to SIP protocol.....................................................................................36
Advance Setting to SIP protocol...............................................................................37
Hot Line Setting .......................................................................................................40
Port Status.................................................................................................................40
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Chapter 6.............................................................................................. 41
System Administrations...................................................................... 41
Management....................................................................................................................41
Save Configuration...................................................................................................41
Access Control..........................................................................................................42
Set To Default Configuration....................................................................................42
System Information Display Function......................................................................43
SNTP Setting Function.............................................................................................43
Syslog setting............................................................................................................43
Capture packets Function .........................................................................................44
Appendix A........................................................................................... 45
Voice communications....................................................................................................45
Concepts: Voice port.................................................................................................45
Sample scenario_1: Peer to Peer GSM termination.................................................46
Sample scenario_2: Enterprise SIP + GSM termination ..........................................49
Appendix B........................................................................................... 52
FAQ ..................................................................................................................................52
Appendix C........................................................................................... 54
Firmware upgrade Requirement and Process .............................................................54
Appendix D........................................................................................... 56
VIP-281GS Specifications..............................................................................................56
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Chapter 1
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Introduction
Overview
With years of Internet telephony and router manufacturing experience, PLANET proudly introduces the
The PLANET VIP-281GS VoIP GSM Gateway is a signal-GSM channel gateway that supports SIP and
H.323 VoIP protocol at the same time. The VIP-281GS provides a total solution for integrating
voice-data network and the Global System for Mobile Communications (GSM).
The VIP-281GS is equipped with both FXS and PSTN interfaces, which gives the gateway a wide
range of potential applications. The VIP-281GS can be installed on a PBX trunk line to enrich its
trunks-GSM and VoIP routes. The PBX is able to have voice communication to either VoIP or GSM
environment by the least costs.
Meanwhile, the VIP-281GS is designed for comfort, ease-of-use with a sophisticated and satisfaction to
customers. The VIP-281GS not only inherits traditions of quality voice communications but the
VIP-281GS also eliminates the human resource of VoIP network deployment. With optimized
H.323/SIP architecture, the VIP-281SG is the ideal choices for P2P voice chat and ITSP cost-saving
solution, but also provides network-converting feature to translate the packet network into traditional
PBX system.
With built-in PPPoE/DHCP/DDNS clients, up to 2 concurrent connections in VIP-281GS, voice
communications can be established from anywhere around the world. The VIP-281GS comes with
intuitive user-friendly and powerful management interface (web/telnet), that can dramatically reduce IT
personnel resource and complete GSM/VoIP deployment in a short time. Plus remote management
capability, administrators can monitor machine/network status or proceed
maintenance/trouble-shooting service via Internet browser or telnet session.
Besides, it provides voice channels status display and optimized packet voice streaming over managed
and public (Internet) IP networks.
Network Features
• Point-to-Point Protocol over Ethernet (PPPoE) Client Support:
The router has a built-in PPPoE client for establishing a DSL link connection with the ISP. There is
no need to install a further PPPoE driver on computers.
• Smart QoS
The smart QoS provides stable voice quality while users access internet from private LAN
to internet at the same time. This device would start suppressing throughput automatically
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when VoIP call was proceeded and it keep full speed access when there is no VoIP traffic.
•
DDNS (Dynamic Domain Name Server)
DDNS is a service that maps Internet domain names to IP addresses. It allows you to provide
Internet users with a domain name (instead of an IP Address) to access Virtual Servers.
• NAT Traversal
The NAT traversal allows gateway to operate behind any NAT/Firewall device. There is no need to
change any configuration of NAT/Firewall like setting virtual server.
VoIP Features
• H.323 / SIP dual mode communication
• SIP 2.0 (RFC3261), H.323v4 compliant
• Peer-to-Peer / H.323 GK / SIP proxy calls
• PSTN lifeline support
• Voice codec support: G.711(A-law /μ-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
• Voice processing: Voice Active Detection, DTMF detection, G.165/G.168 compliant echo canceller,
silence detection.
• Built-in adaptive buffer that helps to smooth out the variations of delay (jitter) in voice traffic.
• Voice channels status display: This function displays each port status such as on-hook, off-hook,
calling number, talk duration, codec.
GSM Features
• SMS Server for SMS sending and receiving
• Worldwide GSM network usable (850/900/1800/1900 MHz)
• Supports GSM PIN code protection
Package Content
The contents of your product should contain the following items:
¾ Voice Gateway VIP-281GS unit
¾ Power adapter
¾ GSM Antenna
¾ Quick Installation Guide
¾ User’s Manual CD
¾ RJ-45 cable x 1
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Physical Details
The following figure illustrates the front/rear panel of VIP-281GS series:
Figure 1-1. Front panel of VIP-281GS
Figure 1-2. Rear panel of VIP-281GS
Front Panel LED Indicators & Rear Panels
Front Panel LED State Descriptions
PWR
WAN Port
Line
Phone
GSM
SMS
On
Off
ON
Flashing
Off
ON
Flashing
Off
On
Flashing
Off
On
Flashing
On
Flashing
GSM GW is powered ON
GSM GW is powered Off
Network connection established
Data traffic on cable network
Waiting for network connection
Line is busy
Ring Indication
Line is not enabled
Telephone Set is Off-Hook
Ring Indication
Telephone Set is On-Hook
GSM Network is found and working properly
Searching GSM Network
Short message waiting Indicator
Sending short message
Table 1-1. Front panel description of VIP-281GS
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Note
The Default WAN IP is http://172.16.0.1. Press RESET button
on rear panel over 5 seconds will reset the VoIP GSM Gateway
to this default LAN/WAN IP address and Username/Password
function.
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Rear Panel Descriptions
A
Phone
Line
SIM
Antenna Connector
WAN
Reset
12V DC (Power)
0 Warning
Phone port was connected to your telephone sets or Trunk Line of PBX.
Can be Connected to PBX or CO line with RJ-11 analog line. PSTN not
FXO port, can’t connect PSTN to VoIP,. When PSTN call comes, it will
transfer to FXS port, let FXS can pick up call from VoIP or PSTN.
The port which you can Insert SIM Card
Connect the antenna to the gateway.
Connect to the network with an Ethernet cable. This port allows your ATA
to be connected to an Internet
ADSL modem, through a networking cable with RJ-45 connectors used on
10BaseT and 100BaseTX networks.
Push this button until 3 seconds, and ATA will be set to factory default
configuration.
The supplied power adapter connects here.
Table 1-2. Rear panel description of VIP-281GS
Incorrectly connecting telephony devices to the RJ11 port
on the Telephony Interface can cause permanent damage to
the VoIP Gateway
ccess device, e.g. router, cable modem,
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Chapter 2
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Preparations & Installation
Physical Installation Requirement
This chapter illustrates basic installation of VIP-281GS series
• Network cables. Use standard 10/100Base-TX network (UTP) cables with RJ45 connectors.
• TCP/IP protocol must be installed on all PCs.
For Internet Access, an Internet Access account with an ISP, and either of a DSL or Cable modem (for
WAN port usage)
Administration Interface
PLANET VIP-281GS provides GUI (Web based, Graphical User Interface) for machine management
and administration.
Web configuration access
To start VIP-281GS web configuration, you must have one of these web browsers installed on
computer for management
•
Microsoft Internet Explorer 6.0 or higher with Java support
Default WAN interface IP address of VIP-281GS is 172.16.0.1. You may now open your web browser,
and insert http://172.16.0.1 in the address bar of your web browser to logon VIP-281GS web
configuration page.
VIP-281GS will prompt for logon username/password, please enter: admin / 123 to continue machine
administration.
:
Figure 2-1. Login prompt of VIP-281GS
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Í
Note
Please locate your PC in the same network segment
(172.16.0.x) of VIP-281GS. If you’re not familiar with
TCP/IP, please refer to related chapter on user’s manual
CD or consult your network administrator for proper network
configurations.
WAN IP address configuration via web configuration interface
Execute your web browser, and insert the IP address (default: 172.16.0.1) of VIP in the adddress bar.
After logging on machine with username/password (default: admin / 123), browse to “WAN Setting”
configuration menu, you will see the configuration screen below:
L
Hint
Figure 2-2. WAN port configuration
Connection Type Data required.
Static IP
DHCP
PPPoE
Table 2-1. WAN port configuration descriptions
Please consult your ISP personnel to obtain proper PPPoE/IP
address related information, and input carefully.
If Internet connection cannot be established, please check
the physical connection or contact the ISP service staff
for support information.
The ISP will assign IP Address, and related information.
Get WAN IP Address automatically; it is no need to
configure the DHCP settings.
The ISP will assign PPPoE username / password for
Internet access,
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Chapter 3
3
Network Service Configurations
Configuring and monitoring your VoIP Gateway from web browser
The VIP-281GS integrates a web-based graphical user interface that can cover most configurations
and machine status monitoring. Via standard, web browser, you can configure and check machine
status from anywhere around the world.
Overview on the web interface of VoIP GSM Gateway
With web graphical user interface, you may have:
More comprehensive setting feels than traditional command line interface.
Provides user input data fields, check boxes, and for changing machine configuration settings
Displays machine running configuration
To start VIP-281GS web configuration, you must have one of these web browsers installed on
computer for management
Microsoft Internet Explorer 6.0 or higher with Java support
Manipulation of VoIP GSM Gateway via web browser
Log on VoIP GSM Gateway via web browser
After TCP/IP configurations on your PC, you may now open your web browser, and input
http://172.16.0.1 to logon VoIP GSM gateway web configuration page.
VoIP gateway will prompt for logon username/password: admin / 123
Figure 3-1. Login prompt of VIP-281GS
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Figure 3-2. System configuration
VIP-281GS Setup for Quick Start
System Configuration
After finishing the authentication, the Main menu will display 3 parts of configuration, please click
“Advance Setup” to enter advance configuration:
1. Network Setup (WAN Port Type Setup)
For most users, Internet access is the primary application. The Gateway support the WAN interface for
Internet access and remote access. The following sections will explain more details of WAN Port
Internet access and broadband access setup. When you click “WAN Setting” from within the Advance
Setup, the following setup page will be show.
Figure 3-3. WAN setting
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Three methods are available for Internet Access
Fixed IP User
If you are a leased line user with a fixed IP address, fill out the
following items with the information provided by your ISP.
IP Address
Netmask
Default Gateway
check with your ISP provider
check with your ISP provider
check with your ISP provider
Table 3-1. WAN setting descriptions
ADSL Dial-Up User (PPPoE Enable)
Some ISPs provide DSL-based service and use PPPoE to establish communication link with end-users.
If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE. If
they do, you need to select this item.
Figure 3-4. PPPoE enable setting
Three methods are available for Internet Access
User Name
Password
Confirm Password
Enter User Name provided by your ISP
Enter Password provided by your ISP
Enter Password to confirm again
Table 3-2. PPPoE enable descriptions
DHCP Client (Dynamic IP): (Get WAN IP Addre ss automatically)
IP Address: If you are connected to the Internet through a Cable modem line then a dynamic IP
address will be assigned.
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Figure 3-5. DHCP setting
2. VoIP Basic Setup:
STEP1 : Configure VoIP Call Signal Protocols :
User could select H.323 or SIP Protocol, and click “select”
Figure 3-6. FXS/GSM number setting
STEP2 : Configure the numbering with Phone(FXS)/GSM ports.
FXS Number
GSM Number
STEP3: Let GW Register to Gatekeeper/SIP Proxy Server
(If user does not have Gatekeeper/SIP Proxy Server, Please go to STEP 4: Outgoing Dialing Plan)
Gatekeeper IP address
SIP Proxy Server IP
addresses
STEP 4: Outgoing Dialing Plan
The representation number is the phone number of the telephone
that is connected to Phone port
The representation number is the phone number of SIM CARD
Table 3-3. FXS/GSM number descriptions
There is a gatekeeper address fields. If this gateway does
not want to register to any gatekeeper, just set value
0.0.0.0 to the primary gatekeeper address.
There is a SIP Proxy Server address fields. If this gateway
does not want to register to any SIP Proxy Server, just set
value 0 .0.0.0 to the sip proxy server address.
Table 3-4. Gatekeeper/SIP proxy descriptions
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The purpose of “Outgoing Direct Call” setting is to let user create a proprietary dialing plan when this
Gateway is not registered to any H.323 Gatekeeper or any SIP Proxy Server. This setting can also
assign some dialing plan to local ports (including prefix strip, prefix addition).
Through this setting, user can directly map a number to a specific gateway (IP address).
Figure 3-7. Dial plan setting
In the “Outgoing Dial Plan” settings:
“Leading Number” is the leading digits of the dialing number.
“Min Length” and “Max Length” is the min/max allowed length you can dial.
“Strip Length” is the number of digits that will be stripped from beginning of the dialed number.
“Prefix Number” is the digits that will be added to the beginning of the dialed number.
“Destination” is the IP address of the destination Gateway that owns this phone number.
STEP 5: Finishing the Wizard Setup
After completing configuration setup, please press “Save Configuration” and “Reboot” hyperlinks to
save the configuration and rebooting Gateway. After 20 Seconds, you could re-login the Gateway.
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Chapter 4
GSM Setup
GSM Setup
In GSM Setup, VIP-281GS provides user the major parts GSM function to configure:
GSM Setup Label
GSM Parameter allows you to modify the option of GSM
GSM Parameter
network.
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PSTN Dialplan
GSM Dialplan
Send SMS
Receive SMS
Terminate Black List
Originate Black List
Users could apply any dial policy by setting Dial Plan to route the
Calls to PSTN
Users could apply any dial policy by setting Dial Plan to route the
Calls to GSM Network.
The Option is used to send short message to mobile phones
This function is used to save the short messages on SIM card to
a external file
The numbers in the list can not call from VoIP to GSM Network
The numbers in the list can not call from GSM Network to VoIP
Table 4-1. GSM setup descriptions
Figure 4-1. GSM setup setting
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GSM Parameter
Figure 4-2. GSM parameter setting
GSM parameter configuration
PIN Code Protection
Failsafe Mechanism
Baby Call
FXS Battery Reverse
Talking Time limit
GSM Frequency Select the GSM band
CLI Presentation
Enable PIN Code protection
If enable, when GSM Network is failed or GSM Gateway is out
of the GSM service range. ALL the calls from FXS will route to
PSTN port.
When the calls come to FXS port, it will call hot line number to
GSM automatically.
Enable battery reverse generator.
The period of talking time, when the time ends, a beep sound
will come out as a warning sound.
If disable this option, the phone number of SIM card won’t be
shown in the callee side.
If enable, PSTN and GSM number will be carried over Internet
CLI Detection
in p2p call and asterisk server. if the version of asterisk is old
then 1.4,please enable asterisk 1.3.
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Answer Supervision
Support Battery Reverse Detection.
GSM Receive Gain
GSM Transmit Gain
GSM Answer Mode
VOIP TO GSM Hot Line
It’s able to adjust the GSM Receive Gain, range from -10db to
6db.
It’s able to adjust the GSM Transmit Gain, range from 30db to
42db.
1. Auto Answer Mode (Default Setting): GSM Port answers
the call once it starts to ring.
2. Connecting Answer Mode:
Case A: “Hot Line Number” was NOT assigned in the
GSM port. GSM answer the call once it starts to ring.
Case B: “Hot Line Number” was assigned and the Hot
line number belongs to remote VoIP device.
In this case, GSM port will not answer (off-hook) the call
till the user picks up the call.
(Note: This case can avoid charging for the call when the
remote VoIP device still ring.)
When VoIP call comes to GSM port, the GSM gateway can
dial out to GSM network automatically with specific phone
number.
Table 4-2.
GSM parameter
descriptions
PSTN Dialplan
PSTN Route Numbers: The numbers which are filled in the form will go through the PSTN line
unconditionally. You can use x as wild card.
Figure 4-3. PSTN dialplan setting
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For examples:
Emergent calls, like 911
Zone Numbers, like 02x (the phone numbers start with 02)
GSM Dialplan
GSM Numbers: The numbers which are filled in the form will go through GSM Network unconditionally.
You can use x as wild card.
Figure 4-4. GSM dialplan setting
For examples:
09x All telephone numbers start with 09
0919x All telephone numbers start with 0919
Send SMS
Figure 4-5. SMS sending setting
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SMS sending configuration
Sending Number
SMS Content
The telephone number which an short message is sent to.
The SMS Content will be sent to the preset telephone
number. If the SMS text is blank, an empty SMS is sent.
The Maximum capacity is 40 characters.
Table 4-3. SMS sending descriptions
Receive SMS
This function is used to save the short messages on SIM card to a external file.
Figure 4-6. SMS Receive Backup setting
Terminate Phonebook
Terminate Phone Book: The following phonebook can set to block or allow. When set to block, call
from VoIP to GSM Network match the phone book will be block. When set to allow, only the phone
number match the phone book will be allow.
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Figure 4-7. Terminate Black setting
Originate Phonebook
Originate Phonebook: The following phonebook can set to block or allow. When set to block, phone
number match phonebook can not call from GSM Network to VoIP, When set to allow, only phone
number match phonebook call allow to make call.
Figure 4-8. Originate Black setting
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Chapter 5
Advance Setup
Network Setup
In Network Setup, VIP-281GS provides user the major parts Network function to configure:
Figure 5-1. Network setup setting
5
Dynamic DNS
DDNS is a service that maps Internet domain names to IP addresses. DDNS serves a similar purpose
to DNS: DDNS allows anyone hosting a Web or FTP server to advertise a public name to prospective
users. Unlike DNS that only works with static IP addresses, DDNS works with dynamic IP addresses,
such as those assigned by an ISP or other DHCP server. DDNS is popular with home network, who
typically receive dynamic, frequently-changing IP addresses from their service provider. To use DDNS,
one simply signs up with a provider and installs network software on their host to monitor its IP address.
Figure 5-2. DDNS date setting
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Three methods are available for Internet Access
User Name
Password
Domain Name
DNS Server IP
Input your DDNS User Name
Input your DDNS Password
Input you set from your DDNS
Input your DNS Server IP
Table 5-1. DDNS date descriptions
Network Management
Network Parameter allows you to modify the access port of gateway.
For example: Setting HTTP port: 80 and Setting TELNET port: 23
Figure 5-3. Access port service setting
VoIP Setup
GSM Gateway support 2 VoIP protocol - H.323 / SIP, you can register to H.323 Gatekeeper or SIP
proxy server. Gateway is n ot a softswitch, it only can use 1 VoIP protocol (SIP/H.323) at the same
time! If you don’t register GK or Proxy server, you can make Peer to Peer call by IP address or domain
name (Setting Dialing plan).
In VoIP Setup, VIP-281GS provides user the major parts VoIP functions to configure:
VoIP Setup Label
The PLANET series gateway support 2~24 phone/line for SIP and
VoIP Basic
H.323 VoIP call applications. You can configure these ports from
this menu.
Dialing Plan
Advanced Setting
Users could apply any dial policy by setting Dial Plan including
outgoing dial plan and incoming dial plan.
VIP-281GS support for silence compression, DTMF Relay, Codec
Selection, FAX mode Option.
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H323 Register Type and H.323 Fast-Start/Normal-Start function.
FXO AC impedance, Volume Adjustment, RRQ TTL, RFC2833
Payload, IP TOS..., etc.
Hot Line Setting
Port Status
Let user can set up “hotline” to dial the phone number
automatically.
Display the telephone interface status.
Table 5-2. VoIP setup descriptions
Figure 5-4. VoIP setup setting
VoIP Basic Configuration to H.323 protocol
Gateway H.323 protocol support H.323 (v2/v3/v4), H.225, Q.931, H.245 and RTP/RTCP. Don’t support
H.235 security, can’t use H.235 security Authentication Username / Password. H.323 protocol is not
good at pass NAT/Firewall; the best way is installed gateway on Public IP Address when it uses H.323.
Configure the numbering with FXS/GSM ports.
Figure 5-5. E.164 number setting
E.164 number setting
The representation number is the phone number of the
FXS Number
telephone that is connected to FXS port.
GSM Number
The representation number is the phone number of SIM
CARD
Table 5-3. E.164 number descriptions
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Configure the ANI (Answer Number Indication) / Caller ID of the FXS/GSM ports
ITSP needs ANI for authorization when gateway calls Off-Net call to PSTN number or mobile phone
number.
Figure 5-6. Caller ID setting
Register to H.323 Gatekeeper
Note: If user does not have Gatekeeper, please go to H.323 Dialing Plan Policy for more
understandings.
H.323 Parameters Label
H.323 ID
Primary Gatekeeper IP
Address
Secondary Gatekeeper
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Figure 5-7. H.323 parameter setting
Sets the unique name of this Gateway, that is communicated as
part of H.323 messaging.
There are two gatekeeper address fields, one is primary, the other
secondary. If this gateway does not want to register to any
gatekeeper, just set value 0 to the primary gatekeeper address. If
the primary gatekeeper address is not 0, the gateway will register
to the primary gatekeeper. If the second gatekeeper is not 0, the
gateway will try to register to the second gatekeeper when failed
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IP Address
Primary Gatekeeper
to register to primary gatekeeper, i.e. if both the primary
gatekeeper and second gatekeeper
Domain Name
Secondary Gatekeeper
Domain Name
H.323 Gatekeeper ID
Voice Cap Prefix
RAS Port Adjustment
Q.931 Port Adjustment
H.323 Call Pass through NAT
H.323 ID
Let user use Domain Name of H.323 Gatekeeper.
The Gatekeeper ID; usually do not need to set this field unless the
gatekeeper must need this value.
Let user set prefix number in RRQ nonstandard voicecap entry.
In H.323 standard the RAS default port number is 1719. The VoIP
gateway provides user to change RAS port number to meet the
network environment.(Some area carrier blocks or forbidden the
default port number)
In H.323 standard the default Q.931 port number is 1720. The
VoIP gateway provides user to change Q.931 port to meet the
network environment. (Some area carrier blocks or forbidden the
default port number)
Sets the unique name of this Gateway, that is communicated as
part of H.323 messaging.
1. Disable : The Gateway operates in public IP address
2. Auto Detection: When the Gateway register to GNU
H.323 Pass Through
NAT method
Gatekeeper, please select this option.
3. Manual Setting: When the Gateway registers to H.323
Gatekeeper and operate under NAT (enable DMZ), please select
this option and key in IP address.
Table 5-4. H.323 parameter descriptions
Dialing Plan to H.323 protocol
The “Dialing plan” needs setting when the user uses the method of Peer-to-Peer H.323 VoIP call or
registering H.323 Gatekeeper mode. The H.323 Dialing Plan has two kinds of directions: Outgoing (call
out) and Incoming (call in).
Peer-to-Peer call mode: Effective
Outgoing Dial Plan
Incoming Dial Plan
Registering to H.323 Gatekeeper mode: Effective
Peer-to-Peer call mode: Effective
Registering to H.323 Gatekeeper mode:
The leading number would register to H.323 Gatekeeper
Table 5-5. Dial plan descriptions
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In the “Outgoing Dial Plan Configurations” settings: Maximum Entries : 50
Outbound number
Length of Number
Delete Length
Prefix no.
Destination IP / DNS
The leading digits of the call out dialing number.
It has two text fields need filled: “Min Length” and “Max Length” is
the min/max allowed length you can dial.
The number of digits that will be stripped from beginning of the
dialed number.
The digits that will be added to the beginning of the dialed number.
The IP address / Domain Name of the destination gateway that
owns this phone number.
Table 5-6. Outgoing dial plan descriptions
Figure 5-8. Outgoing dial plan setting
Scenario description: Normally dial
001x leading call out, call to destination IP address: 172.16.0.100
002x leading call out, call to destination domain name: h323gw.test.com
Figure 5-9. Outgoing dial plan setting
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Scenario description: Speed dial
If user dials “101”, the gateway automatically dials “1234567890” to destination IP address:
172.16.0.101
If user dials “202”, the gateway automatically dials “0987654321” to destination IP address:
172.16.0.202
Figure 5-10. Outgoing dial plan setting
In the “Incoming Dial Plan Configurations” settings: Maximum Entries : 50
Inbound number
Length of Number
Delete Length
Prefix no.
The leading digits of the dialing number.
It has two text fields need filled: “Min Length” and “Max Length” is
the min/max allowed length you can dial.
The number of digits that will be stripped from beginning of the
dialed number.
The digits that will be added to the beginning of the dialed number.
Table 5-7. Incoming dial plan descriptions
Figure 5-11. Incoming dial plan setting
Scenario description: Termination call to GSM for one-shoot call
GSM Port: SIM card was connected to GSM Gateway and standby for incoming/outgoing calls
properly.
H.323 leading number “081x” incoming, and delete the first one digit “0”, and call to GSM number.
Note: “081x” will be registered to H.323 Gatekeeper if “Register to GK” was enabled, show as below:
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Figure 5-12. Incoming dial plan setting
Advance Setting to H.323 protocol
In Advanced Setting , VIP-281GS provides user three major parts function to configure:
One is “VoIP Advance”, the other are “Telephone Advance” , “Network Adv ance” and “Tone Table
Setting”
Advance Setting
H.323 VoIP Advance Configurtion
After the VoIP call is connected, when you dial a digit, this digit is sent
to the other side by DTMF tone. There are two methods of sending
the DTMF tone. The first is “in band”, that is, sending the DTMF tone
DTMF Relay for H.323
in the voice packet. The other is “out band”, that is, sending the DTMF
tone as a signal. Sending DTMF tone as a signal could tolerate more
packet loss caused by the network. If this selection is enabled, the
DTMF tone will be sent as a signal.
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Figure 5-13. VoIP Advance setting
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This selection could force the Gateway to use normal start mode
H.323 Mode
H.323 H.245 T unneling
H.323 Registration type
H.323 RRQ TTL
(default mode) or fast start mode when establishing a VoIP call. Many
other gateways only support normal start mode, enable this selection
when it is necessary. The default is disabled (using fast start mode).
This selection could force the Gateway to use H.245 Tunneling when
establishing a VoIP call The default is disabled (using fast start
mode).
There are 2 choices for this setting. “Gateway” means it will act as the
VoIP gateway. “Terminal” means it will act as the IP phone terminal.
This command configures the number of seconds that the gateway
should be considered active by the H.323 Gatekeeper. The gateway
transmits this value in the RRQ message to the gatekeeper.The
default value is “0”.
When a VoIP call is incoming, the Gateway will ring a specific phone
set. The H.323 call signaling part could be connected or alerting
during this ringing period. If this selection is enabled, the H.323
signaling part is connected during the ringing period. The benefit of
this situation is that the remote side could hear the status of the
H.323 Autoanswer
MAC Authentication
Watchdog
specific port. That is, the remote side will hear ring back tone if the
Gateway is really ringing the phone set. If the phone set is busy, the
remote side will hear busy tone. The disadvantage of this situation is
that the H.323 connected time is not the real voice call connected
time. So, if billing is recorded for this Gateway, this function should be
disabled.
Some Gatekeeper register need UA send MAC address to
Authentication, you need enable this function.(Default is disable).
When your gateway shutdown, or something happen that made
gateway can’t work fine. Watchdog will reboot your gateway
automatically when it can’t work.
Table 5-8. VoIP Advance descriptions
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Telephone Advance
Figure 5-14. Telephone Advance setting
H.323 T eleph one Advance Configuration
Silence Compression
(VAD)
Voice Codec option
Dial Complete Tone
Dial Termination key
FXS Impedance
Phone (Line) in/out
volume:
If this function is enabled, when silence is occurred for a period of
time, no data will be sent across the network during this period in
order to save bandwidth.
(If you use Asterisk, please disable Silence Compression, it maybe
make you call disconnect.)
The codec is used to compress the voice signal into data packets.
Each codec has different bandwidth requirement. There are four
kinds of codec, G.723, G.729AB, G.711_u and G.711_A. The
default value is G.723.
When you use the VoIP call, you will hear “DuDu” voice that is dial
complete tone. If you don’t want to hear that tone, you can disable
it. (Default is enabling).
Setting Termination key to speed up VoIP dial. Select “*” or “#” to
Termination key.
The FXS provides 600/900 OHM impedances for selection.
You can adjust the Phone (Line) in/out volume, range from -9db to
9db
(If you adjust too bigger, maybe generation some ECHO or noise)
Ring Frequency
DTMF tone power
You can configure how long the Ring Frequency do you want to
use.
Sometimes you input DTMF, but no request. You can adjust this
function, range from -6db to +6db.
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Ring Frequency
You can configure how long the Ring Frequency do you want to
use.
Enable battery reverse to detect polarity from PSTN line. The
FXO Battery Reverse
PSTN line can send H.323 case: Sending the Q.931 connect
signal to caller when detecting polarity reverse from PSTN line.
Table 5-9. Telephone Advance descriptions
Network Advance
Figure 5-15. Network Advance setting
H.323 Netwrok Advance Configuration
G.723/G.729
Bandwidth
IP TOS
Setting G.723 / G.729 voice compression size. Quality and Packet
size can adjust by you want.
Enable / Disable Type of Service in IP packets.
Table 5-10. Network Advance descriptions
VoIP Basic Configuration to SIP Protocol
Gateway SIP support SIP(RFC3261), SDP(RFC2327), RFC2833, STUN(RFC3489), Symmetric RTP,
outbound proxy, ENUM(RFC2916),and RTP/RTCP.SIP NAT pass through Function can support 80%
NAT/Firewall that you don’t setting DMZ/Virtual server in router or Firewall.
Select “SIP Protocol”
SIP number (username) and Password Setting: Please fill out the SIP account including username /
password from ITSP.
Note: Support digits and character base SIP Account / username, some SIP Server use character
username to login, and a number to call number (ie. VoIPBuster), if your servers don’t support this,
number/Account is the same, please input the same username, and now only support digits type for
SIP number / username
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Figure 5-16. Port number setting
Port Number / Password Setting
Input SIP number (Username), if your server support account
Number
and number (different), input the number, else number/account
are the same username.
Reg
Let your sip account register SIP Server, click this option.
Input SIP account (Username), if your server support account
Account
and number (different), input the number, else number/account
are the same username.
Password
Input Password that ITSP support.
This allows gateway can use single SIP account for multiple
Use Public Account
ports. User input the only one account in port one field for
registering the ITSP.
Table 5-11. Network Advance descriptions
SIP Hunting Table: This allows gateway can answer SIP call from internet by Hunting.
For example: Port 1 and port 2 is hunting for the port 1 SIP account. If the port 1 is incoming call, the
other one SIP call from internet will ring port 2.
Figure 5-17. SIP hunting table setting
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Figure 5-18. SIP proxy setting
SIP Proxy Server Setting
Domain/Realm
SIP Proxy Server
Register Interval
(seconds)
SIP Authentication
Outbound Proxy Server
SIP NAT Traversal Method
Enter the SIP realm in this field
Enter the SIP service IP address or domain name in this field
(the domain name that comes after the @ symbol i n a full
SIP URI).
This field sets how long an entry remains registered with the
SIP register server. The register server can use a different
time period. The gateway sends another registration request
after half of this configured time period has expired.
Enable or disable MD5 authentication with SIP proxy server.
The outbound proxy method is just very like the proxy server
built-in NAT pass-through solution, except that the packets
need to pass through the outbound proxy server.
STUN client / Symmetric RTP
Table 5-12. SIP proxy descriptions
Figure 5-19. NAT pass setting
If your gateway under the NAT/Firewall, you should setting different NAT Pass function. if you setting
STUN/Outbound Proxy, you should have a STUN/Outbound proxy server. If they can’t pass NAT or
one way talk happen, try to open “DMZ” and virtual server “5060” port in router.
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NAT Pass Setting
NAT Pass Method
Default use Symmetric RTP pass function.
Setting your STUN server information, default STUN server is
STUN Client
FWD STUN server.
Outbound Proxy
Setting your Outbound Proxy server information.
Support
Local SIP Port
Setting local use SIP port, default is 5060.
Table 5-13. SIP proxy descriptions
Dialing Plan to SIP protocol
The “Dialing plan” needs setting when the user uses the method of Peer-to-Peer or registering SIP
proxy server mode. The SIP dialing plan has two kinds of directions: Outgoing (call out) and incoming
(call in).
Outgoing Dial Plan
Incoming Dial Plan
Peer-to-Peer call mode: Effective
Registering to SIP Proxy Server Mode: Effective
Peer-to-Peer call mode: Effective
Registering to SIP proxy server mode: The leading number would
register to SIP proxy server
Table 5-14. Dialing plan descriptions
Figure 5-20. Outgoing dial plan setting
In the “Outgoing Dial Plan Configurations” settings: Maximum Entries : 50
Outbound number
The leading digits of the call out dialing number.
It has two text fields need filled: “Min Length” and “Max Length” is
Length of Number
the min/max allowed length you can dial.
Delete Length
Prefix no.
The number of digits that will be stripped from beginning of the
dialed number.
The digits that will be added to the beginning of the dialed number.
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Destination IP / DNS
The IP address / Domain Name of the destination gateway that
owns this phone number.
Destination SIP Port
It is the UDP port of the remote SIP proxy, which usually refer to
the SIP server on the ITSP side.
Table 5-15. Outgoing dial plan descriptions
Figure 5-21. Incoming dial plan setting
In the “Incoming Dial Plan Configurations” settings: Maximum Entries : 50
Inbound number
The leading digits of the dialing number.
It has two text fields need filled: “Min Length” and “Max Length” is
Length of Number
the min/max allowed length you can dial.
Delete Length
The number of digits that will be stripped from beginning of the
dialed number.
Prefix no.
The digits that will be added to the beginning of the dialed number.
Table 5-16. Incoming dial plan descriptions
Advance Setting to SIP protocol
In Advanced Setting, VIP-281GS provides user three major parts function to configure:
One is “VoIP Advance”, the other one is “Telephone Advance” , “Network Advance ” and “Tone T able
Setting”
VoIP Advance
Figure 5-22. VoIP Advance setting
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SIP VoIP Advance Configurtion
After the VoIP call is connected, when you dial a digit, this digit is sent to
the other side by DTMF tone. There are three methods of sending the
DTMF tone. The first one is “in band”, that is, sending the DTMF tone in
DTMF Relay for SIP
RFC2833 Payload
Watchdog
Telephone Advance
the voice packet. The second one is “RFC2833”, that is, sending the
DTMF tone as a RTP payload signal. The third one is “SIP Info”, that is,
sending the DTMF tone as a SIP signal. Sending DTMF tone as a
signal could tolerate more packet loss caused by the network. If this
selection is enabled, the DTMF tone will be sent as a signal.
Adjust RFC2833 DTMF payload value; range from 96 to 127, default is
101.
When your gateway shutdown, or something happen that made
gateway can’t work fine. Watchdog will reboot your gateway
automatically when it can’t work.
Table 5-17.
VoIP Advance
descriptions
Figure 5-23. Telephone Advance setting
SIP Telephone Advance Configuration
If this function is enabled, when silence is occurred for a period of time,
Silence Compression
(VAD)
no data will be sent across the network during this period in order to
save bandwidth. (If you use Asterisk, please disable Silence
Compression, it maybe make you call disconnect.)
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The Codec is used to compress the voice signal into data packets.
Voice Codec option
Dial Complete Tone
Dial Termination key
FXS Impedance
FXO AC Impedance
Phone (Line) in/out
volume
Ring Frequency
DTMF tone power
Each Codec has different bandwidth requirement. There are four kinds
of Codec, G.723, G.729AB, G.711_u and G.711_A. The default value is
G.723.
When you use the VoIP call, you will heard “DuDu” voice that is dial
complete tone. If you don’t want to heard that tone , you can disable
it.(default is enable).
Setting Termination key to speed up VoIP dial. Select “*” or “#” to
Termination key.
The FXS provides 600/900 OHM impedances for selection.
The FXO provides wild and complex ac termination impedances for
selection.
You can adjust the Phone (Line) in/out volume, range from -9db to 9db.
(If you adjust too bigger, maybe generation some ECHO or noise)
You can configure how long the Ring Frequency do you want to use.
Sometimes you input DTMF, but no request. You can adjust this
function, range from -6db to +6db.
Table 5-18. Telephone Advance descriptions
Network Advance
Figure 5-24. Network Advance setting
SIP Netwrok Advance Configuration
If this function is enabled, when VoIP call is occurred, the other data will
Smart-QoS
be automatically reduced traffic which across the internet in order to
guarantee the voice bandwidth.
You can configure your bandwidth what the Max byte of download and
Bandwidth control
upload of ADSL modem rate.
G.723/G.729 Bandwidth
IP TOS
Setting G.723 / G.729 voice compression size. Quality and Packet size
can adjust by you want.
Some Router support TOS(Type of Service), when you enable the TOS
function, the router will process those packets firstly.(default is disable)
Table 5-19. Network
Advance
descriptions
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Hot Line Setting
You can set hot line. When the call incoming the hot line port, it will call hot line number automatically.
The hot line calls the number via VoIP, so you setting the hot line number must VoIP number. Usually,
you want to incoming GSM calls transfer to FXS, you only setting the GSM hot line to FXS number.
Port number: Input FXS/GSM wants to call hot line number. The call will via VoIP, so the number must
be the VoIP number.
Figure 5-25. Hot line setting
Port Status
Each of port show status table. You can view all port status. Like on/off hook, caller/callee IP, duration,
and packet loss.
Port Status Display: This selection will display concurrent call status of this gateway. The status
information of each voice channel includes codec, dialing number and destination IP address. The
status is refreshed every 3 seconds.
Figure 5-26. Port status
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Management
Save Configuration
Chapter 6
6
System Administrations
Management Label
You can save configuration and restart the gateway with the default
configuration or with the current running configuration.
Access Control
Set to Default
System Information
SNTP Setting
Syslog Setting
Capture Packets
Users can sets/changes the administrator password...
You can restart the VIP-281GS with the default configuration.
Display software version, WAN Type, VoIP status, VoIP codec, and
phone interface and system information.
SNTP (Simple Network Time Protocol) configuration for
synchronizing gateway clocks in the global Internet.
VIP-281GS can send log information to Syslog Server by UDP ports
514.
The VIP-281GS supports packets capture and save the packets to
your PC.
Table 6-1. Management descriptions
Figure 6-1. Management setting
Save Configuration
This page allows you to click “Save Configuration and Reboot” to save configuration and begin to
restart.
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Figure 6-2. Save setting
Access Control
Changing the Administrator/Guest Password
For security reasons, we strongly recommend that you set an administrator/password for the router. On
first setup the router requires no password. If you don’t set a password the router is open and can be
logged into and settings changed by any user from the local network or the Internet.
Click Access Control Setup, the following screen will open.
Administrator username/password: admin/123
Guest username/password: guest/guest
Figure 6-3. Access control setting
Set To Default Configuration
If you want to reboot the router using factory default configuration, click “Apply” then reset the
router’ s settings to default values.
Figure 6-4. Set to default setting
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System Information Display Function
Click System Information Display to open the Online Status page. In the example, on the foll
owing page, both PPPoE connections is up on the WAN interface, H323/SIP Status, MAC addr
ess, Register Status.., etc.
Figure 6-5. System information
SNTP Setting Function
Click SNTP setting to open the Online Status page. In the example, on the following page:
Figure 6-6. SNTP setting
Use SNTP Setting— when checked, gateway uses a Simple Network Time Protocol (SNTP) to set the
date and time. The gateway synchronizes the gateway’s time after you select the time zone. Use SNTP
Setting; select the time zone which gateway was at.
Syslog setting
Use Syslog server to record your VIP-281GS log file. To set the Syslog server IP address for this
function. Kindly please download for this FREE service at http://www.kiwisyslog.com/index.php
more understandings.
for
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Figure 6-7. Syslog setting
Figure 6-8. Syslog topology
Capture packets Function
Use “Capturer Packets” to record VIP-281GS packets. Users can start and stop the capture then save
the file to PC. Use the Ethereal Tool (www.ethereal.com
Figure 6-9. Capture packets setting
) to analyze the packets.
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Appendix A
Voice communications
The chapter shows you the concept and command to help you configure your PLANET VIP-281GS
through sample configuration. And provide several ways to make calls to desired destination in
VIP-281GS. In this section, we’ll lead you step by step to establish your first voice communication
via web browsers operations.
Concepts: Voice port
There are two type of the voice port, Phone (FXS, Foreign exchange Station) on the printing of the
RJ-11 port, and GSM on the printing of the SIM port, you should find that.
z
Phone port
The Phone port allows the connection to an end node, like telephone, or out-line of PBX system.
Phone port is as like your local phone service provider who provides a number to you. It is easy to
tell that after you have connected an end-device to Phone port and you will hear the dial-tone from
Phone port once the hand set off-hook.
Caution
0
The Phone port is with voltage and current. DO NOT connects the port to any
PBX extension line or PSTN line. This may make the Phone port or your PBX
extension port malfunction.
FXS
412-1111
Figure A-1. Phone port topology
222
or
z
GSM port
The GSM port allows can be inserted a SIM card that already has a fixed number; say 0912-111111.
So the only connections for GSM port will be to your local PSTN or GSM network.
With your GSM connect to GSM network; the Internet Voice can then have a GSM call through this
line/number (0912-111111). Or, locally, you can have an Internet Call through the line 0912-111111.
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Your PBX users will need to know this number in the future.
GSM
0912-111111
222
GSM
Figure A-2. GSM port topology
Sample scenario_1: Peer to Peer GSM termination
In the following samples, we’ll introduce the Peer to Peer GSM termination applications.
In this example, there are two VIP-281GS calling by IP address directly, both VIP-281GS have
inserted the GSM SIM cards into SIM slots, the GSM number are 09127788(GSM_1) and
09583344(GSM_2).
The VoIP number of VIP-281GS_A are ext.100 (FXS) and ext.200 (GSM), the VoIP number of
VIP-281GS_B are ext.300 (FXS) and ext.400 (GSM)
Figure A-3. Peer to Peer GSM topology
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achine configuration on the VIP-281GS:
M
STEP 1:
Pleas
e log in VIP-281GS_A via web browser, browse to the Advance Setup -> VoIP Basic
menu and set the VoIP number as 100 and 200, the sample configuration screen is shown
below:
Figure A-4. VoIP basic settings
STEP 2:
e browse to the Dial Plan menu and add the outgoing dial plan for calling to Pleas
VIP-281GS_B, the sample configuration screen is shown:
Figure A-5. Outgoing dial plan settings
STEP 3:
e browse to the GSM Setup -> PSTN Dial plan menu and set the PSTN outgoing Pleas
number, the sample configuration screen is shown:
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Figure A-6. PSTN Routing table
STEP 4:
Please browse to the GSM Dial plan menu and set the GSM outgoing number, the sample
configuration screen is shown:
STEP 5:
Repeat the same configuration steps on VIP-281GS_B.
Figure A-7. GSM Routing table
Test the scenario:
A. FXS_1 call to GSM_4
1. FXS_1 pick up the telephone.
2. Dial the ext.400 to GSM port of VIP-281GS_B, and get the dial tone.
3. Dial the GSM number #09581122 to establish the voice communication with GSM_4.
B. GSM_3 call to FXS_2
1. GSM_3 dial the GSM number #09127788 to GSM_1, and get the dial tone.
2. Dial the ext.300 to establish the voice communication with FXS_2.
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C. FXS_1 call to PSTN_1
1. FXS_1 pick up the telephone.
2. Dial the PSTN number #10125566 to establish the voice communication with PSTN_1.
Sample scenario_2: Enterprise SIP + GSM termination
In the following samples, we’ll introduce the SIP Proxy and GSM termination applications.
In this example, there are two VIP-281GS; the FXS and GSM ports are register to SIP Proxy Server
(IP PBX).
The out-lines of PBX connect with Phone (FXS) ports of VIP-281GS. The extensions of PBX can
make GSM calls via GSM ports of VIP-281GS.
Figure A-8. Enterprise GSM Routing table
Machine configuration on the VIP-281GS:
STEP 1:
Please log in VIP-281GS_A via web browser, browse to the Advance Setup -> VoIP Basic
menu, set the VoIP registration number as 100/ 200 and the registration server address, the
sample configuration screen is shown below:
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Figure A-9. Port number settings
Figure A-10. SIP proxy settings
STEP 2:
Because the VIP-281GS have registered to IP PBX, all the VoIP calls will send to IP PBX, so
that don’t need to set the dial plan settings.
Figure A-11. Outgoing dial plan settings
STEP 3:
Please browse to the GSM Dial plan menu and set the GSM outgoing number, the sample
configuration screen is shown:
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STEP 4:
Figure A-12. GSM Routing settings
Repeat the same configuration steps on VIP-281GS_B.
Test the scenario:
A. ext.501 call to GSM_3
1. Ext.501 picks up the telephone, and input the trunk code 8 to connect with FXS port of
VIP-281GS_A.
2. Dial the GSM number #09125566 to establish the voice communication with GSM_3.
B. ext.501 call to GSM_4
1. Ext.501 picks up the telephone, and input the trunk code 8 to connect with FXS port of
VIP-281GS_A.
2. Dial the ext.400 to GSM port of VIP-281GS_B, and get the dial tone.
3. Dial the GSM number #09581122 to establish the voice communication with GSM_4.
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Appendix B
FAQ
Q1: What is the default administrator password to login to the gateway?
A: By default, your default username is “admin”; default password is “123” to login to the
router. For security, you should modify the password to protect your gateway against hacker
attacks.
Note: Default guest login username/password: guest/guest
Q2: I forgot the administrator password. What should I do?
A: Press the Reset button on the rear panel for over 5 seconds to reset all settings to default
values.
Q3: What is the default IP address of the router?
A: The default WAN IP address is 172.16.0.1 with subnet mask 255.255.0.0.
Q4: What is different [set to default] and [Factory set to default]?
A: Factory set to default, you must push RST button until 5 second, and gateway will clear
all your setting, and let gateway Wan port become the factory default (172.16.0.1). When
you use setting to default by Web or telnet, it will clear all your setting, but the wan port
setting will be saved. If you remote the gateway, after set to default, you can login gateway
again. No reset the gateway wan port again.
Q5: Why can I call out when the gateway under the NAT?
A: VoIP product almost has NAT Pass through problem. By SIP, there are many NAT Pass
through Function can solve 80% NAT Problem. You can choose STUN/Outbound Proxy/
Symmetric RTP to Pass through NAT, you don’t set any other setting (DMZ/Virtual Server)
by router side. If you use STUN/Outbound Proxy, you must have a STUN/Outbound Proxy
Server to support. If they can’t pass NAT, please open the DMZ/Virtual Server by
Router/NAT/Firewall.
Q6: Why does the one way talk happen?
A: Generally, one way talk happen when use the different codec between VoIP devices
make call. Please check and setting the same codec, most one way talk will be solved.
Q7: Why can I call out by Gateway?
A: Please chick your Gateway is registered SIP Proxy Server (ITSP), and chink your
Internet works fine. Gateway can’t make a call without Internet or SIP Account that from
ITSP supply. You must have a SIP account or know the other Gateway IP/Domain Name,
and then you can make a VoIP call.
Q8: Why I use asterisk by G.729 sometimes disconnect happen?
A: In asterisk setting VAD must disable, if you open Silence Compression (VAD), it will
make call disconnect happen, please disable the option when you use the asterisk.
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Q9: Why can I register and use after setting?
A: After setting, please save configuration and reboot, after reboot you can use new
configuration.
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Appendix C
Firmware upgrade Requirement and Process
1. Environment Requirement
a) A PC with FTP Server (Server-U software)
b) A PC or Notebook witch connected to LAN port of gateway.
c) Put the image (firmware) named “FW-VIP281GS_vxxx.bin ” at the assigned folder in FTP Server.
For example: “FW-VIP281GS_v305.bin” is version 3.0.5L
Note: Free FTP server: 172.16.0.101
username: xxxx, password: xxxx
Environment Architecture (Gateway and FTP server are in Internet):
Figure C-1. Firmware upgrades topology
2. Upgrading Process
a) Notebook Telnet GSM GW -> open DOS mode ->C:> telnet 172.16.0.1 (Default WAN port IP)
b) Please insert login password: 123, and select [4] Upgrade Software
Figure C-2. Main menu
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c) Please input IP address of FTP server like as: 172.16.0.101, username: xxxx, passswd: xxxx, and
image name: FW-VIP281GS_v305.bin
d) Upgrade (y/n): y, then will write the firmware to flash.
e) After writing flash, Please reboot the Gateway.
f) If the new firmware (image) was most different with the previous version, please push the hardware
reset bottom to set to default.
g) If the GSM Gateway is in remote site, please use WEB configuration to set to default.
Figure C-3. Upgrade firmware procedures
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Appendix D
VIP-281GS Specifications
Product H.323/SIP VoIP GSM Gateway
Model VIP-281GS
Hardware
WAN 1 x 10/100Mbps RJ-45 port
FXS 1 x RJ-11 connection
PSTN 1 x RJ-11 connection
GSM 1 x SIM connection
Protocols and Standard
H.323 v2/v3/v4 and SIP (RFC 3261)
SDP (RFC 2327)
Symmetric RTP
Standard
Voice Codec G.711(A-law /μ-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
Voice Standard
Protocols
Advanced Function Smart QoS, IP TOS (IP Precedence) / DiffServ
Network and Configuration
Access Mode Static IP, PPPoE, DHCP
Management Web, Telnet
LED Indications
Dimension (W x D x H) 180 x 110 x 25 mm
Operating Environment 0~40 degree C, 0~90% humidity
Power Requirement 12V DC
EMC/EMI CE, FCC Class B
STUN (RFC3489)
ENUM (RFC 2916)
RTP Payload for DTMF Digits (RFC2833)
Outbound Proxy Support.
Voice activity detection (VAD)
Comfort noise generation (CNG)
G.165/G.168 Echo cancellation
Dynamic Jitter Buffer
SIP 2.0 (RFC-3261), H.323, TCP/IP, UDP/RTP/RTCP, HTTP, ICMP,
ARP, PPPoE, DNS
System: 1, PWR
WAN: 1, LNK/ACT
Line: 1, In-Use/Ringing
Phone: 1, In-Use/Ringing
GSM: 1, In-Use/Standby
SMS: 1, Transmission
Page 57
EC Declaration of Conformity
For the following equipment:
*Type of Product : 2-Port H.323 / SIP VoIP GSM Gateway
*Model Number : VIP-281GS
* Produced by:
Manufacturer‘s Name: Planet Technology Corp.
Manufacturer‘s Address: 11F, No 96, Min Chuan Road
Hsin Tien, Taipei, Taiwan, R. O.C.
This product, which has been issued the test report listed as above in QuieTek Laboratory, is
based on a single evaluation of one sample and confirmed to comply with the requirements of the
following CE/LVD (Low-Voltage Directive; 73/23/EEC) standard.
73/23/EEC relating to electrical equipment designed for use within certain voltage limits and the
Amendment Directive 93/68/EEC.
ESD EN 61000-4-2
RS EN 61000-4-3
EFT/ Burst EN 61000-4-4
Surge Test EN 61000-4-5
CS EN 61000-4-6
Voltage Disp EN 61000-4-11
EMC (R&TTE, Article 3.1b)
EN 301 489-1 V1.6.1
EN 301 489-7 V1.3.1
Radio spectrum (R&TTE, Article3.2)
EN 301 511 V9.0.2 selection of Test-cases form 3GPP
TS 51.010 V7.3.1
Responsible for marking this declarati o n i f the:
⌧ Manufacturer Authorized representative established within the EU
Authorized representative established within the EU (if applicable):
Company Name: Planet Technology Corp.
Company Address: 11F, No.96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C
Person responsible for making this declaration
Name, Surname Jonas Y ang
Position / Title : Product Manager
Taiwan
20 Oct, 2007
Place Date Legal Signature
PLANET TECHNOLOGY CORPORATION
e-mail: sales@planet.com.tw http://www.planet.com.tw
11F, No. 96, Min Chuan Road, Hsin Tien, Taipei, Taiwan, R.O.C. Tel:886-2-2219-9518 Fax:886-2-2219-9528
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