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VGW-402
Quick Installation Manual
12 pgs1.35 Mb0
User Manual
78 pgs2.96 Mb0
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Planet VGW-402, VGW-400FS, VGW-400FO User Manual

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Internet Telephony Gateway
VGW-400 Series
4-Port SIP VoIP Gateway
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Internet Telephony Gateway
VGW-400 Series
Copyright
Copyright© 2013 by PLANET Technology Corp. All rights reserved. No part of this publication may be reproduced, transmitted, transcribed, stored in a retrieval system, or translated into any language or computer language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical, manual or otherwise, without the prior written permission of PLANET. PLANET makes no representations or warranties, either expressed or implied, with respect to the contents hereof and specifically disclaims any warranties, merchantability or fitness for any particular purpose. Any software described in this manual is sold or licensed "as is". Should the programs prove defective following their purchase, the buyer (and not this company, its distributor, or its dealer) assumes the entire cost of all necessary servicing, repair, and any incidental or consequential damages resulting from any defect in the software. Further, this company reserves the right to revise this publication and to make changes from time to time in the contents hereof without obligation to notify any person of such revision or changes. All brand and product names mentioned in this manual are trademarks and/or registered trademarks of their respective holders.
Disclaimer
PLANET Technology does not warrant that the hardware will work properly in all environments and applications, and makes no warranty and representation, either implied or expressed, with respect to the quality, performance, merchantability, or fitness for a particular purpose. PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims liability for any inaccuracies or omissions that may have occurred. Information in this User’s Manual is subject to change without notice and does not represent a commitment on the part of PLANET. PLANET assumes no responsibility for any inaccuracies that may be contained in this User’s Manual. PLANET makes no commitment to update or keep current the information in this User’s Manual, and reserves the right to make improvements to this User’s Manual and/or to the products described in this User’s Manual, at any time without notice. If you find information in this manual that is incorrect, misleading, or incomplete, we would appreciate your comments and suggestions.
Trademarks
The PLANET logo is a trademark of PLANET Technology. This documentation may refer to numerous hardware and software products by their trade names. In most, if not all cases, these designations are claimed as trademarks or registered trademarks by their respective companies.
CE mark Warning
This is a class B device, in a domestic environment; this product may cause radio interferen ce, in which case the user may be required to take adequate measures.
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Federal Communication Commission Interference Statement
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Part 15 of FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television recep tion, which ca n be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
1. Reorient or relocate the receiving antenna.
2. Increase the separation between the equipment and receiver.
3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
4. Consult the dealer or an experienced radio technician for help.
FCC Caution:
To assure continued compliance (example-use only shielded interface cables when connecting to computer or peripheral devices). Any changes or modifications not expressly approved by the party responsible for compliance could void the user’s autho rity to operate the equipment. This device complies with Part 15 of the FCC Rules. Operation is subject to the Following two conditions: (1) This device may not cause harmful interference, and (2) this Device must accept any interference received, including interference that may cause undesired operation.
R&TTE Compliance Statement
This equipment complies with all the requirements of DIRECTIVE 1999/5/EC OF THE EUROPEAN PARLIAMENT AND THE COUNCIL OF 9 March 1999 on radio equipment and telecommunication terminal Equipment and the mutual recognition of their conformity (R&TTE) The R&TTE Directive repeals and replaces in the directive 98/13/EEC (Telecommuni cations Terminal Equipment and Satellite Earth Station Equipment) As of April 8, 2000.
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VGW-400 Series
WEEE Caution
To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment, end
users of electrical and electronic equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately.
Safety
This equipment is designed with the utmost care for the safety of those who install and use it. However, special attention must be paid to the dangers of electric shock and static electricity when working with electrical equipment. All guidelines of this and of the computer manufacture must therefore be allowed at all times to ensure the safe use of the equipment.
Customer Service
For information on customer service and support for the Planet Product, please refer to the following Website URL: http://www.planet.com.tw
Before contacting customer service, please take a moment to gather the following information:
• Internet Telephony Gateway System serial number and MAC address
• Any error messages that displayed when the problem occurred
• Any software running when the problem occurred
• Steps you too k to re solve the problem on your own
Revision
User’s Manual for PLANET Internet Telephony Gateway Model: VGW-400 Series Rev: 1.0 (October, 2013)
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TABLE OF CONTENTS
Chapter 1 Introduction.................................................................................................... 7
1.1 Features.............................................................................................................................10
1.2 Package Contents..............................................................................................................12
1.3 Physical Specifications .....................................................................................................12
1.4 Specifications....................................................................................................................14
Chapter 2 Installation Procedure................................................................................. 20
2.1 W eb Login.........................................................................................................................20
2.2 Configuring the Network Setting......................................................................................21
2.3 Changing IP Address or Forgotten Admin Password........................................................22
Chapter 3 Device Setting.............................................................................................. 23
3.1 Network Configuration..................................................................................................... 23
3.2 Device Time Setting..........................................................................................................25
3.3 Device Advance Setting....................................................................................................27
3.4 User Login Setting ............................................................................................................27
3.5 Debug Setting....................................................................................................................28
3.6 Event Notice......................................................................................................................29
3.7 Auto Provisioning .............................................................................................................29
3.8 SNMP................................................................................................................................30
3.9 PABX Mode......................................................................................................................31
Chapter 4 NAT Setting .................................................................................................. 33
4.1 DHCP Srv.(DHCP Server) ................................................................................................33
4.2 UPNP (Universal Plug and Play server)............................................................................33
4.3 Bandwidth (Bandwidth Control).......................................................................................33
4.4 URL Filter.........................................................................................................................37
4.5 IP Filter..............................................................................................................................38
4.6 MAC Filter........................................................................................................................38
4.7 APP Filter..........................................................................................................................39
4.8 Port Filter..........................................................................................................................39
4.9 Port Fwd............................................................................................................................39
Chapter 5 VoIP Setting.................................................................................................. 40
5.1 SIP.....................................................................................................................................40
5.2 Audio.................................................................................................................................41
5.3 Tone...................................................................................................................................42
5.4 NAT Traversal...................................................................................................................42
Chapter 6 VoIP Advance............................................................................................... 44
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6.1 SIP.....................................................................................................................................44
6.2 SIP.....................................................................................................................................47
6.3 Ring...................................................................................................................................48
Chapter 7 Dialing Plan.................................................................................................. 49
7.1 General..............................................................................................................................49
7.2 Dialing Rule......................................................................................................................49
7.3 Digit Manipulation............................................................................................................50
7.4 Phone Book.......................................................................................................................51
Chapter 8 FXS Setting .................................................................................................. 53
8.1 FXS Line...........................................................................................................................53
8.2 SIP Proxy..........................................................................................................................56
8.3 Caller ID............................................................................................................................57
8.4 Others................................................................................................................................58
Chapter 9 FXO Setting.................................................................................................. 59
9.1 FXO line............................................................................................................................59
Chapter 10 SIP Trunk.................................................................................................... 62
10.1 Create SIP Trunk.............................................................................................................62
Chapter 11 Route Plan.................................................................................................. 66
11.1 For PABX Mode interface...............................................................................................66
11.2 For Non-PABX Mode interface.......................................................................................70
Chapter 12 Status.......................................................................................................... 74
12.1 Device Status...................................................................................................................74
12.2 Line Status.......................................................................................................................74
12.3 SIP Trunk Status..............................................................................................................75
Chapter 13 Maintenance............................................................................................... 76
13.1 Firmware Update.............................................................................................................76
Appendix A - Default Setting ....................................................................................... 77
Appendix B - Changing IP Address or forgotten admin password........................... 78
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Chapter 1 Introduction

Cost-effective, High-performance PoE VoIP Phone
To build high-performance VoIP communications at a low cost, PLANET now introduces the latest member of its gateway family, the VGW-400 Series enterprise-class 4-port SIP VoIP Gateway. The VGW-400 Series provides added flexibility during migration to Unified Communications by supporting the traditional analog devices. These devices include analog phones, fax machines, modems, voicemail systems, and speakerphones. It helps the company to save money on long-distance calls; for example, the remote workers can dial in through a Unified VoIP Communication System just like an extension call but no long-distance call charge would occur. The VGW-400 Series also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.
Standard Compliance
The VGW-400 Series supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VGW-400 Series is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services.
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VGW-400 Series
Enhanced, Full-Featured
The VGW-400 Series is a full-featured enhanced business SIP Gateway that addresses the communication needs of the enterprises. It provides the FXO and FXS gateway with SIP protocol IP device which allows connection with PSTN telephone line and with analog telephone set to make or receive V oIP call over Internet or VPN network. This device is suitable for office PABX to enable to have VoIP call without changing cabling, dial plan and extension number.
The VGW-400 Series supports all kinds of SIP-based gateway features and multiple contact filter functions, such as 4 SIP trunk accounts, both IPv6 and IPv4 protocols, flexible dial plan and route plan features, and switch analog and VoIP signal to help both protocols to communicate.
Business Gateway
Secure, High
It can effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS
-Quality VoIP Communication
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(Quality of Service), 802.1Q VLAN tagging, and IP TOS (Type of Service) technology. Using voice and data VLAN can easily separate the data and voice, thus maintaining the best quality.
Supporting Caller ID
Both the FXS and FXO ports of the VGW-400 Series support caller ID function, help u ser identify calling number easily and verify number. It also helps to block anonymous call by filtering strange calls. The FXS port transmits Caller ID, while the FXO port receives Caller ID. The Caller ID interoperates with analog phones, public switched telephone networks (PSTN) and private branch exchanges (PBXs).
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1.1 Features

¾ Highlights
Supports SIP 2.0 (RFC3261) Supports IPv6 and IPv4 simultaneously Up to 4 SIP service domains and Caller ID Supports auto HTTP provision and fax feature Flexible Routes Plan, Dial Plan and SIP Trunk Life-line for emergency calls
¾ Internet Features
IPv4 (RFC 791) and IPv6 IPv6 auto configuration (RFC 4862) IPv6 only, IPv4 only or dual stack
VGW-400 Series
MAC clone setting Vendor Class ID DDNS ( Planet DDNS, Easy DDNS, DynDNS) DNS client Firewall URL / IP / MAC / Port Filter Port forwarding (TCP, UDP or both) Bandwidth control (download and upload), maximum bandwidth priority setting
¾ SIP Applications
SIP Session Timer (RFC 4028) SIP Session Refresher: UAC or UAS SIP Encryption Supports Outbound Proxy / STUN NAT Traversal Supports Primary and Backup SIP Server
¾ Call Features
Supports peer to peer dialing 2-line FXO connects to PSTN line 2-line FXS connects to analog phone set or PABX. Caller ID recognition DTMF (before/after 1st ring) and FSK (before 1st ring ), ETSI
and Bellcore
DTMF Caller ID start and stop BIT configurable T.38 fax volume configuration
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¾ FXO/FXS Line Configuration
Line ID / Line Phone number Polarity Reversal detection or generation for call establish and billing VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing Outgoing SIP Caller ID selection Caller ID detection mode by country selection
¾ Routing Plan
Prefix match and length Priority / Cyclic / Simultaneous Ring Programmable Hunting Cycle
VGW-400 Series
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1.2 Package Contents

Thank you for purchasing PLANET Internet Telephony Gateway system, VGW-400 Series. This Quick Installation Guide will introduce how to finish the basic setting of connecting the web management interface and the Internet. Open the box of the Internet Telephony Gateway system and carefully unpack it. The box should contain the following items:
z VGW-400 Series x 1 z Quick Installation Guide x 1 z User’s Manual CD x 1 z Power Adapter x 1 (12V) z RJ-45 x 1
If any of above items are damaged or missing, please contact your dealer immediately.

1.3 Physical Specifications

¾ Dimensions
Dimension 175 × 32 ×126 mm Net weight 500g (with package)
Front Panel of the VGW-400 Series
Rear Panel of the VGW-400 Series (VGW-402)
Rear Panel of the VGW-400 Series (VGW-400FS)
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Rear Panel of the VGW-400 Series (VGW-400FO)
LED definitions
LED definitions
LED Function Description
VGW-400 Series
Power
Status Proxy
WAN
LAN
Port 1 - 4
When the power adapter is connected, the LED will light up
green.
When system startup successfully, the LED will light up green.
When the gateway is registered successfully to a SIP Proxy, this
will light up green.
This LED lights up green when the gateway’s WAN port is
physically connected to the public internet. When data is
transmitted through this port, it will flash green.
This LED lights up green when the gateway’s LAN port is
physically connected to a local network (Refer to Rear Panel
section). When data is transmitted through this port, it will flash
green.
The status LED for FXO and FXS ports, these LED light up amber
orange when connected phone is engaged in a conversation
mode (FXO). It will flash amber orange when there is an
incoming call (FXS).
Port Function Description Reset
FXS Ports
Press and hold over 5 seconds to reload factory default setting,
this action will erase all existing settings configured on this
gateway.
The status LED for FXS port, it will light up amber orange when
the connected phone’s handset is lifted, or when the connected
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phone is engaged in a conversation. It will flash amber orange
when there is an incoming call.
FXO Ports
LAN
WAN
DC 12V
The status LED for FXO port. When there is no PSTN line
connected, this LED becomes blinking to remind you. When
PSTN line is connected and no talking, the LED is OFF. When a
line is using, the LED becomes steady light up.
10/100 Base-T RJ-45 socket for LAN port, connects to PC for
management purpose.
10/100 Base-T RJ-45 socket for WAN port, connects to wide
area network.
The power socket, input AC 100V~240V; output DC12V, 2A
Button Action Description
VGW-400 Series
Reset

1.4 Specifications

Product
Hardware
WAN
LAN
Voice
Press less than 5 secs Press over 5 secs
System reboot. Reset to Factory Default
Please be reminded to reset to factory default. Uploaded music setting (on hold music) and backup file will not be removed.
VGW-400 Series
1 x 10/100Mbps RJ-45 port
1 x 10/100Mbps RJ-45 port
4 x RJ-11 connection
(VGW-402: 2 x FXS, 2 x FXO)
(VGW-400FS: 4 x FXS)
Protocols and Standard
Data Networking
(VGW-400FO: 4 x FXO)
IPv4 (RFC 791) and IPv6
IPv6 auto configuration (RFC 4862)
IPv6 only, IPv4 only or dual stack
MAC address (IEEE 802.3)
MAC clone setting
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Vendor Class ID
IP/ICMP/ARP/RARP/SNTP
Static IP
DHCP Client (RFC 2131), WAN port
DHCP Server, LAN port
NAT Server (RFC 1631)
PPPoE Client / DNS Client / TFTP Client
DDNS (Planet DDNS, Easy DDNS, DynDNS)
Firewall
URL / IP / MAC / Port Filter
Application Program Filter
Port Forwarding (TCP, UDP or both)
Bandwidth control (download and upload), maximum bandwidth priority
setting
UPnP Server at LAN port
Voice Gateway
Behind NAT, use DMZ for NAT traversal
SNTP with time zone and Daylight Saving
TCP/UDP (RFC 793/768), RTP/RTCP (RFC 1889/1890), IPV4 ICMP (RFC
792)
VoIP VLAN Support 802.1Q, 802.1P
VLAN ID Range: 2 to 4094
VLAN Priority: 0 to 7 (Highest Priority)
QoS: DiffServ (RFC 2475), TOS (RFC791, 1394)
RFC3261 compliance
Supports up to 4 SIP Trunks to Register
SIP UDP Protocol
Supports SIP compact Form
Supports SIP HOLD Type: Send Only, 0.0.0.0 or inactive
SIP Session Timer (RFC 4028)
SIP Session Refresher: UAC or UAS
SIP Encryption
MD5 Digest Authentication (RFC2069/RFC2617)
Reliability of provision response PRACK (RFC3262)
Early/Delay Media support
Offer/Answer (RFC3264)
Message Waiting Indication (RFC3842)
Event Notification (RFC3265)
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REFER (RFC3515)
Supports Outbound Proxy
Supports Primary and Backup SIP Server
Supports STUN NAT Traversal
Supports “rport” parameter (RFC 3581)
Configure SIP local Port
SIP QoS Type: DiffServe or QoS Accept Proxy Only : Yes or No
G.711 A-law/μ-law, G.729A, G.723.1 (6.3K, 5.3K)
Select voice codec priority : Local or Remote
Voice Payload size (ms) configuration
Silence Suppression
VAD/CNG
LEC : Line Echo Canceller
Audio Codec
Functions
Max Echo Tail Length (G.168): 32, 64 and 128ms
Packet Loss Compensation
Automatic Gain Control
In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
Adaptive/Configurable Jitter Buffer
G.168 Acoustic Echo Cancellation
Configure RTP basic Port
RTP QoS Type : DiffServ or TOS
Phone Book ( 50 records ) for peer to peer calls
Dialing Plan with drop, replace, Insert dialing digits
Selects first digit and inter digit timeout duration (Sec)
Selectable Call Progress Tone Support Specified Line Calling
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Call Functions
Internet Telephony Gateway
VGW-400 Series
Supports Peer to Peer dialing
FXO connects to PSTN Line
FXS connects to analog phone set or PABX.
Caller ID recognition DTMF (before/after 1st ring) and FSK (before 1st
ring ), ETSI and Bellcore
DTMF Caller ID start and stop BIT configurable
Current Drop Detection to release FXO port
Disconnect tone recognition to release FXO port
Tone Generation: Ring Back, Dial, Busy, call waiting, ROH, Warning,
Holding, Stutter dial tone and disconnect tone
Configure Tone Frequency, Cadence, Level and Cycle
Select Tone specification by Country name List
Global Country Based Tone Specification
NAT Traversal support STUN, UPNP and Behind NAT
Out-Band DTMF with RFC2833 and SIP Info
FXO/FXS Line Configuration
RFC2833 Payload type: 101 or 96
DTMF send out ON and OFF Time configure
DTMF incoming recognition Minimum ON and OFF time
DTMF Relay Volume configuration
T.38 FAX Volume configuration
Flash Time transmit via SIP Info (Enable or Disable)
Message Waiting Indication (Stutter Tone Notice)
Blocks Anonymous Call
Call Hold , Call Transfer
Activates or deactivates : Line ID, Line Phone number
Polarity Reversal detection or generation for call establish and Billing
HOT Line to desired phone number
Plays voice file to incoming call
Repeats playing voice file counts
Self-recorded voice files to upload
Generates FLASH TIME to PSTN network
T.38 or FAX Relay Type
Incoming and outgoing dB value configurable
Dialing Answer Delay time to establish call path
Answers PSTN incoming call after how many ring cycles
Caller ID detection mode by Country selection
VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
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Outgoing SIP Caller ID Selection
Supports 4 SIP Trunk Accepts desired SIP Proxy incoming calls Only
Flexible Routing Plan
Flexible Dial Plans
FXS Analog 2-wire interface
Prefix Match and Length
Priority Ring
Cyclic Ring
Simultaneous Ring
Programmable Hunting Cycle
Backup Routes with Digit Manipulation
Default Routes
Retrieves transfer call from 3rd party by dial code (default: *#)
Inter digit time out setting
First digit dial out delay time setting
End of dial keypad number
Dial Rule : Match dial prefix and maximum digits length ( 1-15 )
Phone Book can be exported or imported
Flash Time Detection: range from 80 to 800 ms
ON-HOOK Voltage -48Vdc
FXO Analog 2-wire interface
Management
Configure Ring Cadence, Frequency and Voltage
Supports Polarity reversal for Billing
Service Up to 1 Kilo-meter distance to analog telephone set
Generate Current Drop Time (Open Loop Disconnect time)
Incoming Ring frequency recognition range: 10 to 70 Hz
Incoming Ring ON time recognition range: 0 to 8000ms
Incoming Ring OFF time recognition range: 0 to 8000ms
Incoming Ring Level recognition range: 10 to 95Vrms
Flash Time Detection: range from 80 to 800 ms
Configure Ring Cadence, Frequency and Voltage
Administrative Telnet CLI and HTTP, HTTPS
HTTP provision through MAC address
Multilingual Web User Interface
3 Levels of User Access Right with Password protection with different Web
Language (Administrator, Supervisor and User)
HTTP/HTTPS Service Access limitation from WAN port
Configure Service ports at HTTP, HTTPS and telnet Services
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Phone Debug Module: Device Control, Call Control, DB, Verbose
SIP Debug Module: Register, Call, SIP Message, Others
SNTP Debug Module
Device Debug Module
DSP Debug
Provides System Status Logs
Connect to external SYSLOG Server
Status display: Network, Line, SIP Trunk status
Diagnostics (debug through Syslog Event Notice)
Debug in real time by Telnet
Auto Provision via HTTP Server
SNMP V2/Trap
Configuration Backup/Restore
Dual Firmware Image Backup
Reset to factory Default
Environments
Power Requirements
Operating Temperature
Operating Humidity
Weight
Dimensions (W x D x H)
Emission
Connectors
12V DC, 2A
0 ~ 45 degrees C
10%~90% relative humidity, non-condensing
550g
175×32×126 mm
CE, FCC, RoHS
Two 10/100 BASE-T RJ-45 Ethernet ports
Four RJ-11 ports
DC power jack
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Chapter 2 Installation Procedure

2.1 Web Login

Step 1. Connect a computer to an LAN port on the VGW-400 Series. Your PC must set up to
the same domain as 192.168.0.X as the VGW-400 Series
Step 2. Start a web browser. To use the user interface, you need a PC with Internet Explorer
(version 6 and higher), Firefox, or Safari (for Mac).
Step 3. Enter the default IP address of the VGW-400 Series: 192.168.0.1 into the URL
address box.
Step 4. Enter the default user name admin and the default password admin, and then click
Login to enter Web-based user interface.
(Default IP)
Default WAN IP Default subnet mask Default Gateway Default PC IP Default Login User Name Default Login Password
172.16.0.1
255.255.255.0
172.16.0.254
192.168.0.1 admin admin
Login page of the VGW-400 Series
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For security reason, please change and memorize the new password after this first setup.

2.2 Configuring the Network Setting

Step 1. Go to Device Setting → Network
VGW-400 Series
Network setting page
Step 2. Edit your WAN port IP information.
There are three types of IP Support. They are IPV4 Only, IPV4 / IPV6, IPV6 Only. There are also three types of WAN port connection. They are Static IP, PPPoE (Point-to-Point Protocol over Ethernet), DHCP. You can find detailed setting process in the user manual.
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Selection of IP Support / Network Connection Type

2.3 Changing IP Address or Forgotten Admin Password

To reset the IP address to the default IP Address “192.168.0.1” (WAN) or reset the login password to default value. Press the reset button on the front panel for more than 5 seconds After the device is rebooted, you can login the management WEB interface within the same subnet of 192.168.0.xx.
Reset Button
After pressing the “Reset” button, all the system data will be reset to default; if possible, back up the config file before resetting.
.
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Chapter 3 Device Setting

From this setting category, all devices related to parameters can be found here.
Network Configuration

3.1 Network Configuration

Parameter Descriptio Setting:
z IP Support: IP stack to be supported (IPV6 and IPV4 or IPV6 or IPV4 only)
WAN Setting:
1 Network Type:
2 IP Address
3 Net mask
4 Default Gateway
DHCP Tag
5
(option 60)
DHCP Tag
6
(option 61)
n:
support “Fixed IP”; ”DHCP”; ”PPPoE”
IPV4 address IPV4 network subnet mask IPV4 Default gateway Input Vendor class identifier or not.
Input Client identifier or not.
7
IPV6 Network Type
Auto configuration or manual configuration
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8 IPV6 IP Address
9 IPV6 IP Gateway
IPV6 IP Prefix
10
Length
11 DNS Server1
12 DNS Server2
11 VOIP VLAN
VOIP VLAN ID
12
(2-4096)
LAN Setting:
IPV6 address IPV6 default Gateway IPV6 prefix length
Primary DNS Server IP network Secondary DNS Server IP network Enable VOIP VLAN or not. When enable VOIP VLAN, the W AN port
can be only accessed by VLAN. If it is required to manage the VGW Series Gateway, Administrator can use LAN port to access this gateway instead. VLAN ID range to be used
Management
1
Mode
2 NAT mode
3 IP Address
4 Net mask
5 Bridge mode
This LAN port is used for management purpose, not used for register to SIP Server or data/voice routing.
DHCP function on the LAN port. The LAN port functions as a DHCP server, network devices connected to them will be assigned one IP address according to DHCP server IP range. (Please refer to command “NAT setting” on the left side commands how to define DHCP IP address.) IPV4 address
IPV4 network subnet mask At this mode, both WAN and LAN ports are configured to
Switch/Hub features. LAN port access to WAN port directly.
DNS Setting:
1 DDNS
2 Domain Name
It support Planet DDNS, Easy DDNS, DynDNS or disable the DDNS feature. Input your Domain Name
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3 User Name
4 Password:
For Planet DDNS function more detail information please refers the Appendix: Planet DDNS page.
Input your user name
Input your password

3.2 Device Time Setting

VGW-400 Series support SNTP with time zone and daylight saving.
Device Setting > Time
Parameter Descriptio
1 Current Time
NTP Time
2
Server
NTP Refresh
3
Interval(sec)
4 Time Zone
n:
Current Time, date and year display.
SNTP time server IP address
The interval time to sync NTP server in seconds
The time-zone where VGW Series Gateway is located.
- Standard: Use a predefined standard time zone
- Customize: Use a user defined time zone
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5 Daylight Saving
6 Daylight Bias
7 Daylight Start
8 Standard Start
Auto adjust daylight saving timer or not The offset added to the Bias when the time zone is in daylight
saving time The date that a time zone enters daylight time
- Month: 01 to 12
- Week Day: Sunday to Saturday
- Apply Week (Day:01 to 05, Specifies the occurrence of day
in the month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)
- Hour: 00 to 23
The date that a time zone enters daylight time
- Month: 01 to 12
- Week Day: Sunday to Saturday
- Apply Week (Day:01 to 05, Specifies the occurrence of day
in the month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)
- Hour: 00 to 23
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3.3 Device Advance Setting

Parameter Description:
1
HTTP Service:
The Administrator Web service port (the default is 80)
VGW-400 Series
2
3
4
HTTPS Service:
Telnet Service:
HTTP/HTTPS Service access on WAN
The https web service port (the default is 443) The telnet service port (the default is 23)
When click the disable option;
The WEB service will be rejected on WAN port, so please be careful
with this function. If you wanted to enable WAN port again, you
need to
access this device from its LAN port to connect to WEB pages and
enable
- WAN port.

3.4 User Login Setting

Three level of users can be used, administrator, supervisor, user. Each level of users has
different predefined access level.
Extension Settings
Item Explanation
Administrator
The administrator level user who has full access authority to VGW­Series Gateway .
Supervisor
The supervisor level user who has limited administrative access right.
User
The user access right which only allows setting some user related features.
User ID
Login User ID
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Password Confirm Password Language
Login Password Confirm new password again
The desired web page language used when the account was login.

3.5 Debug Setting

VGW-400 Series provides the real time debug to syslog or through telnet interface. It generates the debug information based on debug level and modules. Since the generating debug will consume system resources, it is recommended to turn on only for necessary and under Planet FAE’s instruction.
Item Explanation
Syslog Check for start from Any Time Syslog Start (YYYY/MM/DD HH:MM) Syslog Stop (YYYY/MM/DD HH:MM)
Enable or disable to send system information to syslog server or not Always send syslog or only during a specified time range.
Always send syslog or only during a specified time range.
The syslog stop sending time
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Syslog Server Syslog Port DSP Debug DSP Capture server DSP Capture port
Syslog server IP address Syslog server service port (default is 514) Enable or disable to send DSP information to capture log syslog capture server IP address
syslog capture server service port (default is 50000)

3.6 Event Notice

VGW- Series Gateway can send Syslog Event Notice when it has the following cases:
1. Register Failure or re-registered
2. FXO RJ-11 cable was plugged or unplug
3. Ethernet reconnected
4. System started
Item Explanation
Syslog Notice Syslog Server Syslog Port
Enable or disable to send system events to syslog server or not Syslog server IP address syslog server service port (default is 514)

3.7 Auto Provisioning

TheVGW-400 Series can be provisioned by HTTP Server for large deployment. Please contact
Planet for availabilities.
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9510: (This feature is not yet available now. Please don’t select at present
Select HTTP:
Item Explanation
Http Config URL Refresh
internal used only interval to check whether have a new configuration/firmware or
not in minutes
interval(minute) User ID Password
specify the Login ID for http authentication specify the password for http authentication

3.8 SNMP

SNMP Agent:
Item Explanation
SNMP Agent Read Only Community Name Read Write Community Name SNMP Agent Access
Enable SNMP or not. The community name to read through SNMP protocol
The community name to read and write through SNMP protocol.
Enable SNMP to be accessed through WAN port or not.
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on WAN
Trusted Peer:
Item Explanation
Type
Any Address: Any address can retrieve the SNMP information. Specify an IP Address: Only the IP address listed can retrieve the
SNMP information. Normally, it will be the SNMP manager’s IP address.
Specify a Subnet: Only the network specified can retrieve the SNMP information.
VGW-400 Series
IP address Subnet Mask
The IP address for a trusted peer The network mask for a trusted peer
SNMP Trap:
Item Explanation
SNMP Trap Destination Community
Enable SNMP trap or not The IP address for SNMP manager to receive the SNMP trap
The communication name for sending the SNMP trap

3.9 PABX Mode

This quick setting is dedicated to be used forVGW-400 Serie s to become an inte r-connection in between PSTN Lines and analog trunk lines from traditional PABX.
When this mode was changed (enables to disable or disable to en able). It will clean all of route plan and recovery to default route.
PABX mode is for VGW-402 Only
The call scenario will be working as follows:
1. For FXO incoming call, it will be routed to corresponding FXS directly (Line1 to TEL1, Line2 to TEL2) For FXS outgoing call, it will be routed to VOIP except those prefix set in FXO
2. dialing Prefix.
3. For VOIP incoming call from Sip Trunk number, it will be routed to FXS based on the called number.
Note: If you are dialing to SIP trunk number, and hear the dial tone from
(For VGW-402 Only)
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VGW-series Gateway. Please check the SIP Trunk configuration. It might be configured to option mode at “1 stage dialing”.
4. When VOIP call is failed to be called out such as register fail (this means registration to proxy accounts are all failure, but not include SIP TRUNK num ber) or network issue, the call will be routed to FXO as backup.
5. When
VGW-400 Series is malfunction, IP network disconnection or power is failu re,
s will be directly bypassed to FXO automatically.
all call
Item Explanation
PabxMode Enable or Disable PABX mode, default is “Enable”.
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Chapter 4 NAT Setting

VGW-400 Series can support NAT, 2 Ethernet ports (management mode) or bridge mode.
Here is the setting for NAT related service.

4.1 DHCP Srv.(DHCP Server)

Item Explanation
DHCP Server Client Range Start
Enable DHCP server or not. Specify DHCP client lease start IP
IP Client Range End IP Default Gateway Submask DNS Server 1 DNS Server 2
Specify DHCP client lease end IP Specify the default gateway Specify the subnet mask. Specify the DNS server 1 addre s s
Specify the DNS server 2 addre s s

4.2 UPNP (Universal Plug and Play server)

Item Explanation UPNP Server
Enable UPNP server or not.
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4.3 Bandwidth (Bandwidth Control)

By using bandwidth control feature, the user can manage the traffic based on their needs.
Bandwidth Control:
Item Explanation
Bandwidth Control Download Bandwidth Upload Bandwidth
Maximum Bandwidth and Reserved Bandwidth:
Setup Method: bandwidth control method, percentage or specify the required bandwidth Percentage: total bandwidth
Item Explanation
priority 1 priority 2 priority 3
Specific
Enable bandwidth control or not. Specify total band width for download (unit: kbps). 0 indicates no
limitation.
Specify total band width for upload (unit: kbps). 0 indicates no limitation.
highest priority percentage Normal priority percentage low priority percentage
Item Explanation
priority 1 – Download priority 2 – Download: priority 3 – Download: priority 1 – Upload
highest priority download bandwidth
normal priority download bandwidth
low priority download bandwidth
highest priority upload bandwidth
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priority 2 – Upload: priority 3 – Upload:
In order to setting which target is belonged to which priority. The following is the setting method for target’s priority.
IP Target
normal priority upload bandwidth low priority upload bandwidth
Item Explanation
Priority Type Configure Type
Priority value for the target
The target type is set to IP
Unique IP or a range of IP addresses
¾ Unique:
IP Address: the IP address to be set
¾ IP Range:
Start IP: The starting IP for a range End IP: The stopping IP for a range
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Port Target
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Item Explanation
Priority Type Configure Type
Priority value for the target
The target type is set to port number
Unique port number or a range of port number
¾ Unique:
Port: the port number to be added Protocol: protocol for the port
¾ Port Range:
Start port: the starting port number End port: the stop port number Protocol: protocol for the port range
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Application Target
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Item Explanation
Priority Type Application
Priority value for the target Application The list for the application
DSCP target
Item Explanation
Priority: Type DSCP
Priority value for the target DSCP value The DSCP will be mapped to the priority
VGW-400 Series supports below firewall features.
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4.4 URL Filter

Item Explanation
URL Filter

4.5 IP Filter

The specified URL will be blocked
VGW-400 Series
Item Explanation
IP Filter Local IP address Protocol
The specified IP address to be blocked The LAN side IP address to be forwarded TCP, UDP or both are used for port forward

4.6 MAC Filter

Item Explanation MAC Filter
The MAC address to be blocked, please follow these formats of picture.
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4.7 APP Filter

Item Explanation APP Filter
application to be blocked
VGW-400 Series

4.8 Port Filter

Item Explanation
Port Filter: Port Range
Protocol
Enable port Filter or not. Starting and stopping port to be forward. If you are using only 1 port,
please set the starting equal to stopping port. TCP, UDP or both are used for port blocked.

4.9 Port Fwd

Item Explanation
Port Fwd Port Range
Protocol Local IP address Local Port
Enable port forward feature or not Starting and stopping port to be forwarded. If you are using only 1
port, please set the starting port equal to stopping port. TCP, UDP or both are used for port forward
The LAN side IP address to be forwarded The LAN side port to be forwarded. If you are using the port range,
this port indicates the starting port.
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Chapter 5 VoIP Setting

5.1 SIP

VGW-400 Series
Item Explanation
Session Timer Session Expires (sec) Min SE
Session Timer Refresh Method PRACK
SIP Local Port SIP Qos Type
Accept Proxy Only
Enable session timer or not (RFC 4028) This is the setting of initial session timer expires time according to
RFC4028 - Session Tim ers in the Session Initiation Protocol.
The minimum session timer allowed when receiving a call with session timer value according to RFC 4028. The session timer refresh method
Enable provision ACK or not (RFC 326 2)
- None: Disable PRACK
- Supported: When select this mode, 100rel will be added to
- Require: PRACK is mandatory required.
The SIP local service port (default is 8080) Quality of Service Type for SIP signaling
- None: Not using QOS Tag and not enables QOS.
- DiffServ: Differentiated Services Value. Input DSCP value
- TOS: Type of Service which include IP precedence value
Only accept the call coming from the SIP proxy. Not accept peer to peer call at this mode.
the support list. It indicates VGW-400 Series can support the PRACK but not mandatory.
0-63 for DSCP and TOS.
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5.2 Audio

Item Explanation
VGW-400 Series
Codec 1~5 G.711u Payload Size G.711a Payload Size G.729 Payload Size G.723.1 Payload Size: Bit Rate
Codec Priority
DTMF Relay
The preference codec priority G.711 u-Law payload size G.711 A-law payload size G.729A p aylo ad size G.723.1 payload size G.723.1 bit rate used
5.3K bit rate is used
6.3K bit rate is used Selection order to match the remotely SDP for codec selection.
Local SDP Order: Use local SDP order to match codec  Remote SDP Order: Use Remotely SDP order to match
codec
In-Band DTMF: use inband DTMF instead of out of band. RFC 2833(fall back to SIP-INFO): Use RFC 2833 if the SDP
negotiation could be done. Or use SIP INFO for DTMF relay.
SIP INFO: Use SIP-INFO DTMF relay
Silence Suppression
RTP Basic Port
RFC 2833(fall back to Inband): Use RFC 2833 if the SDP
negotiation could be done. Or use inband DTMF transmission. Enable: Start the voice activity (silence) detection when detect silence for 60 seconds, it will hang up the call (For FXO use) Disable: Send silence packets as normal voice packet (no silence detection) The RTP starting port. Each channel will be added additional
10. For example, the RTP basic port is 16384, thus call 1 will
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use 16384 while call 2 will use 16394 and etc.
RTP QoS Type:
IP QoS tag for RTP stream
DiffServ: The differentiated service QoS tag will be used.
Input DSCP value 0-63 for DSCP.
TOS: Type of Service which include IP precedence value
and TOS.

5.3 Tone

The setting page is used to setup the tone to be generated (FXS) or detected (FXO). The detected tone is the Disconnect 1 & 2 (for FXO use) and the others are for generating (when
FXS received the “bye” from IP side or waiting time out by analog phone which keeps handset pick up, it will send busy tone to analog phone). To recognize correct disconnect
tone is very important for PSTN status supervision to release FXO port after call was dropped.
Please use Country Template to select your local country profile which will be applied. Click to load those country tone parameters to system and change if it is necessary. For those
countries are not showed in the list, please select a closed co untry and edit tone p arameters to match your country . You can send an email with the tone definition to Planet if you would like to put your country tone into the list.

5.4 NAT Traversal

VGW-400 Series supports the following NAT traversal methods when it was situated behind route
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NA T Traversal:
Item Explanation
VGW-400 Series
Disable STUN (T ype 1,2)
STUN (All)
UPNP
Behind NA T
Disable NAT traversal features Enable STUN for NAT traversal. Since STUN can be used only for type 1 and type 2 NAT servers, it is recommended to use this option. When STUN client detect current NAT is type 3, it stops the STUN feature operation. STUN Server: STUN Server IP address No matter which NAT type server is used, STUN is always to be used for NAT traversal. STUN Server: STUN Server IP address Enable UPnP client for NAT traversal. Please note that the IP sharing box (or Router) needs to support uPnP feature. Use DMZ for NAT traversal IP Sharing Address: publi c IP sharing address. You need to specify the port mapping or DMZ for all required ports
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Chapter 6 VoIP Advance

6.1 SIP

VGW-400 Series
Item Explanation
SIP Hold Type
SIP Compact Form
SIP on hold message sending method. Send Only: Set the SDP media to send only when send an on-hold SIP message.
0.0.0.0: Set the SDP connection to 0.0.0.0 when send an on-hold SIP message. Inactive: Set the SDP media to inactive when send an on-hold SIP message. Enable SIP compact form or not. When enable this feature, the
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connected SIP proxy is required to support compact form.
Session Refresher
SIP T1 (msec)
SIP T2 (msec)
Who will send dialog to keep alive message (re-invite or update). UAC: User Agent Client will do the refresh (default setting) UAS: User Agent Server will do the refresh T1 determines several timers as defined in RFC3261. For example, when an unreliable transport protocol is used, a Client Invite transaction retransmits requests at an interval that st art at T1 seconds and doubles after every retransmission. A Client General transaction retransmits requests at an interval that starts at T1 and doubles until it reaches T2. (Default Value: 500ms) ** Determines the maximum retransmission interval as defined in RFC3261. For example, when an unreliable transport protocol is used, general requests are retransmitted at an interval which starts at T1 and doubles until reache s T2. If a provisional response is received, retransmission continue but at an interval of T2. (Default Value: 4000ms) **
SIP T4 (msec)
Invite Linger Timer
General Linger Timer
Cancel General No
T4 represents the amount of time the network takes to clea r message between client and server transactions as defined in RFC3261. For example, when it works with an unreliable transport protocol, T4 determines the time that UAS waits after receiving an ACK message and before terminating the transaction. (Default Value: 5000ms) ** After sending an ACK for an INVITE final response, a client cannot be sure that the server has received the ACK message. The client should be able to retransmit the ACK upon receiving retransmissions of the final response for this timer. This timer is also used when a 222 response is sent for an incoming Invite. In this case, the ACK is not part of the Invite transaction. After a UAS sends a final response, the UAS cannot be sure that the client has received the response message. The UAS should be able to retransmit the response upon receiving retransmissions of the request based on this timer. When sending a CANCEL request on a General transaction,
Response Time (msec)
the User Agent waits for cancel General No Response Timer milliseconds before timeout termination if there is no response
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for the cancelled transaction(Default Value: 10,000 ms).**
General Request Timeout Timer (msec)
Cancel Invite No Response Timer (msec)
Provisional Timer (msec)
First Response Timer (msec)
After sending a General request, the User Agent waits for a final response general Request Timeout Timer milliseconds before timeout termination (in this time the User Agent retransmits the request every T1, 2*T1,…T2,…milliseconds)** When sending a CANCEL request on an Invite request, the User Agent waits for this timer before timeout termination if there is no response for the cancelled transaction. The provisional Ti mer is set when receiving a provisional response on an INVITE transaction. The transaction will stop retransmissions of the INVITE request and will wait for a final response until the provision T imer was expired. If you set the provision Timer to 0, no timer is set. The INVITE transaction will wait indefinitely for the final response. When sending a request out, the User Agent waits this timer for any response received from UAS. If timer is expired and no any SIP message is received, the User Agent will think the request is failed. The default is 5 seconds.
MWI Subscript Expires (sec)
Line Congestion Code
SIP-Info Flash Mode
Encrypt
You can Enable or Disable the MWI subscribe. The default is 600 sec. If a new voice mail is arrived, the stutter tone will be used instead of regular dial tone. This feature is dedicated to
FXS only.
When receiver’s end was contacted successfully from originated site but the receiver site is busy and does not wish to answer the call at this time, the system will response the code, default is 600. (FXO only) When you enable the feature, system will make flash key to send SIP message by sip-info. Disable: disable encryption function. VGCP is a proprietary layer 2 link proto c ol working at between IP stack and NIC driver for VoIP anti-blocking. The core patent-pending VGCP is industry's most st ate-of-art voice service provider class security protocol whose scalability and flexibility results in not to compromise voice quality and overhead. VGCP controls and monitors full voice signaling and media flow intelligently; meanwhile disguise sip and RTP packets into normal allowed data packets such as DNS and
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TFTP, and makes two-way encryption and decryption driven by user-customized policy. VGCP is fully transparent to upper SIP proxy or UA whi ch means Voice Guard@ can work with any 3rd party soft phone/AT A/Gateway/IP Phone/IADs and SIP Proxy or Server not like some competitors which take effect on their own device and soft switch.

6.2 Audio

The setting page includes the device related to audio settings.
VGW-400 Series
Item Explanation
RFC 2833 Payload Type DTMF Send On Time(msec) DTMF Send Off Time(msec) DTMF Detect Min on
96 or 101. It is recommended to use 101. When generate DTMF, the DTMF ON time will be sent (default value is 70 ms) When generate DTMF, the DTMF OFF time will be sent (default value is 70 ms) The minimum DTMF ON time period will be processed as a
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Time (msec)
DTMF Detect Min off Time (msec)
DTMF Relay Volume T.38 Fax Volume T.38 Redundant Depth
T.38 ECM Min Jitter Buffer (msec) Max Jitter Buffer (msec) Max Echo Tail Length (G.168)
regular DTMF event. A smaller ON time less than this will be ignored. The default value is 60ms. The minimum DTMF OFF time for the same DTMF value. A smaller OFF time less than this and the new DTMF digit is the same as previous one will be handled as 1 digit only (same digit and not a new digit). The DTMF relay volume The T.38 fax relay volume The T.38 redundant packet depth. It could be 0 (no redundant), 1 or 2. It is recommended to set to 2. The T.38 error correction mode. Default value is ON. The minimum delay time of Jitter buffer. The Maximum delay time of Jitter buffer.
Enable the echo cancellation feature. The default setting is “128ms”.
Jitter Opt. Factor
Impedance
Jitter buffer dynamic factor for optimize. Please set to 7 unless under Planet’s instruction to change. Selected analog phone’s impedance. (for FXS port use)

6.3 Ring

The ring cadence, voltage and frequency were configured to the p hone.
Item Explanation
Frequency (10~70HZ) Ring on (0~8000ms) Ring off (0~8000ms) Ring level (10~95volt)
Specify the ringing freque ncy value (default is 20HZ) Specify the ringing on value (default is 1000msec) Specify the ringing off value (default is 2000msec) Specify the ringing level (default is 94 volt RMS value
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Chapter 7 Dialing Plan

7.1 General

Item Explanation
VGW-400 Series
First Digit Time Out
Inter Digit Time Out
End of Digit
Retrieve Number:

7.2 Dialing Rule

Specify the duration of the first digit to be dialed whe n the FXO port was OFF Hook. The range is 1~60 sec. Specify the interval of entering between two digit s. If the interval setting time is expired, the gateway sends out the DTMF digits immediately. The time range is 1~10 sec. The assigned key was treated as end of dial and dial out immediately. it forces the line to retrieve back if VIP-400 series makes a transfer call to 3rd party but it DOES NOT answer and put this call go into voice mail service. You can press the preprogram code to retrieve back this call from transferred 3rd party. Default code is “*#”.
Dialing rule is used to speed up the dialing procedure. Some users don’t like to use the end of dialing digit such as “#”, the administrator can use dialing rule instead. The longest prefix will be matched first.
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Item Explanation
VGW-400 Series
Dialed Prefix Max Digits
The following is an example for dialing rule: Mobile call is starting with 09 and it is 10 digits Long distance call is starting with 0 and it is 10 digits International call is starting with 00 and its max digit should be less than 32
The others are local call and 8 digits
Emergency call is starting with digit “1” and length is 3 digits
The Dialing rule can be set as follows: Prefix, max digits 09, 10 0, 10 00, 15 1, 3 2, 8
The prefix to be matched The digits will be received based on the Dialed Prefix.
3, 8 4, 8 5, 8 6, 8 7, 8 8, 8 9, 8

7.3 Digit Manipulation

The Digit Manipulation (DM) will be processed based on prefix and DM group after the DNIS (Called Party) was determined.
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Item Explanation
VGW-400 Series
DM Group
Matched Prefix
Matched Length
Different DM group have different application as follows.
FXO: This DM group is used for FXO port with 2 sta ges
dialing. Af ter the DNIS (Called party messages) is collected, this DM group will be processed before enter the routing procedure.
FXS: This DM group is used for FXS dialing out.  VOIP: This DM group is used for VOIP incoming call. Af ter
the DNIS is collected in 2 stages dialing or 1 stag e dialing, this DM group will be processed before entering the routing procedure.
1-4: These DM groups are used for backup routing
purpose. When a backup routing is used, the administrator can select a DM group to be processed
before starting the backup routes. The prefix to be matched for DM. The longest prefix will be matched first. Set to 0 to ignore the length. The other 1-32 are the digit length to be matched as a condition.
Start Pos Stop Pos Replace Value
Example of Digit Manipulation Settings:
Prefix Len Start
Pos
886 0 0 0 002 8862123456 0028862123456 886 12 0 0 002 8862123456 8862123456 886 0 2 5 002 8862123456 8800223456 886 0 30 30 002 8862123456 8862123456002 886 0 1 6 8862123456 83456
The start digit position to be replaced. The stop digit position to be replaced. The value to be replaced.
Stop Pos
Replace Value
Test DNIS
(called number)
Result DNIS (dial out called number)

7.4 Phone Book

Phone Book is used for peer to peer call.
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Item Explanation
VGW-400 Series
Name
Tel No
Export Import
This field supports called number only. If you enter words or text here, it wills routes to proxy server automatically .
Enter called number and IP address. Please follow this sample of picture, as the format of “number@uri:port”. (default port is
5060) To backup the phone book records. To reload setting of phone book.
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Chapter 8 FXS Setting

The FXS line setting includes each line number and SIP proxy settings.

8.1 FXS Line

Item Explanation
VGW-400 Series
Line ID State TEL No Hotline TEL
FXS line The line is active or not The telephone number of each FXS port If hot line is enable, this field shows the hot line number
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Item Explanation
Line ID Line Type Line State
Forward reason:
FXS Line number (T1 to T2) FXS or FXO (depend on device model). Set to active if you would like to use this line. Otherwise, set to Inactive. Unconditional forward: forward this call without any
condition.
Busy forward: Forward the call when phone is busy.  No answer forward: forward the call when the call does not
answered after any answer timeout.
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Forward TEL: The forwarding telephone number once
Forward action was activated.
No answer timeout (seconds) Call waiting
Reject Anonymous Call Hot line Hot line TEL
Polarity Reversal generation
Current Drop generation
The no answer timeout will be used (default is 120 sec)
Enable call waiting or not. When disable call waiting features, the second incoming call will be rejected. Reject the anonymous incoming call or not Enable to disable hot line feature The number to be dialed automatically after the user pickup the phone. Enable Polarity Reversal of tip/ring of RJ-11 phone line for FXS as billing signal or not. When a FXS calls to VOIP and answered by the remote party, VGW-400 Series generates reverse signal to FXS as a billing start. When VOIP side disconnects call, VGW-400 Series reverse back as a billing stop signal. Enable current drop (0 voltage) when VOIP is disconnected (Remote party drop the call).
Input(Encode)Gain Output(Decode)Gain FAX Relay
Voice mail subscription Caller ID mode
SIP caller ID mode
Register Type
Adjust the volume from FXS/FXO to IP side (default is 0 dB) Adjust the volume from IP side to FXS/FXO (default is 0 dB) Enable T.38 Fax Relay or T.30 Fax Bypass or not.
(T.30 Fax Bypass only supports G711a law)
Enable voice mail subscription (MWI) or not.
Inhibit: don’t send caller ID to analog phone.  Transparent: send caller ID to analog phone. Inhibit: don’t send caller ID to IP SIP side  Transparent: send caller ID to IP SIP side Register: register to proxy. If it is not registered to SIP
proxy, the FXS line still can use SIP trunk for VOIP call.
Predefine: When it is set to predefine, VGW-400 Series
does not send register message out.
Internal: When it is set to internal, VGW-400 Series does
not send register message out, the FXS line still can use SIP trunk for VOIP call or call locally.
TEL No User ID User Password
The registrar telephone number The SIP user ID for register and call making The SIP password for register and call making
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Display Name
The SIP display name

8.2 SIP Proxy

The SIP proxy server defined here is dedicated used for FXS lines.
Item Explanation
Domain Primary proxy server Primary proxy server port Outbound Proxy server Outbound Proxy server port Primary Proxy server keep Alive
The SIP domain for register or call making Primary SIP registrar server address Primary SIP registrar server port number
Primary outbound proxy server address Primary outbound proxy server port number
Using through NAT and keep the port.
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Keep Alive Time(sec) Secondary Proxy
Secondary proxy server Secondary proxy port Secondary outbound Proxy server Outbound Proxy server port Register Expire: Secondary Proxy server keep Alive Keep Alive Time(sec)
Specify time to send SIP register message to proxy server. Enable secondary proxy or not. When enable it, the primary and secondary proxy will be registered at the same time. Secondary SIP registrar server address
Secondary SIP registrar server port number Secondary outbound proxy server address Secondary
Secondary outbound proxy server port number
SIP register time to leave Using through NAT and keep the port.
Specify time to send SIP register message to proxy server.

8.3 Caller ID

The call ID sends to FXS port of analog phone set to display caller name or phone number.
Item Explanation Caller ID Mode
Caller ID mode to be used for phone (FSK Bellcore, FSK ETSI,
Polarity Reverse before caller ID Dual tone before caller
DTMF) Start polarity reverse to FXS port before sending the caller ID
Send Dual Tone before caller ID (for FSK ETSI used only)
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ID
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VGW-400 Series
Caller ID present
DTMF caller ID start digit DTMF caller ID stop digit
The timing to send the caller ID (Before first ring, after first ring, after first short ring) specify the DTMF caller ID start digit (default is D, the range is A to D or #) specify the DTMF caller ID start digit (default is C, the range is A to D or #)

8.4 Others

Flash time and current drop generation/detection time
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Chapter 9 FXO Setting

The FXO Setting contains the FXO related parameters.
Item Explanation
VGW-400 Series
Line ID State TEL No Hotline TEL

9.1 FXO line

FXO line The line is active or not The reference telephone number (e.g. PSTN TEL of line) If hot line is set, this field shows the hot line number
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Item Explanation
VGW-400 Series
User ID User T ype Line State TEL NO
Polarity Reversal Detection
Current Drop for disconnect:
FXO Line number The line type is FXO Set to active if this Line is activated. Otherwise, set to Inactive. This field can be used as a reference remark for this line. Normally, you can put the connected PSTN line’s phone number here for reference. When enable the Polarity Reversal Detection feature, VGW-400 Series uses the polarity reversal signal once call was established for FXO outgoing call and start to count talking time for Billing purpose. When disable the polarity Reversal Detection, VGW-400 Series uses “Dialing Answer Delay
Time” commands to set time (seconds) to start billing time
once SIP call was established. Use Line current drop as a disconnect supervision to release FXO port. When remote PSTN side user drop call, the local PSTN switch send Current drop signal to FXO port to recognize
Incoming call handling
Playback voice file
Repeat Count
Voice file name
this situation. The call handling policy for a FXO incoming call. Hot line TEL: When a PSTN Line incoming call was
detected and after the FXO Answer this call based on Ring Count Configuration, VGW-400 Series sends SIP call to the specified hot line TEL number through the Route Plan.
2 Stage Dialing: When a PSTN Line incoming call was
detected and after the FXO Answer this call based on Ring Count Configuration, VGW-400 Series answers this call and play either Dial Tone or Voice Greeting file to PSTN side. And wait for PSTN side user to dial number to send to IP
SIP Trunk or FXS ports. To enable playing voice greeting file or not. (Used for FXO port Only) Repeat how many counts to play voice greeting file. (Used for FXO port with 2-Stage Dialing Only ) Specify the file path and file name to upload. Please make sure
(MuLaw-mono 8K)
that the file format needs to be G.711U, 8K, 8 bits raw file.
(Used for FXO port Only)
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Flash Time FAX Relay
Input(Encode)Gain Output(Decode)Gain Dialing Answer Delay Time (sec)
PSTN Answer Ring Count
Flash Time wi ll be sent to PSTN line. Enable T.38 Fax Relay or T.30 Fax Bypass or not.
(T.30 Fax Bypass only supports G711a law)
Adjust the volume from PSTN to IP side (default is 0 dB) Adjust the volume from IP side to PSTN (default is 0 dB) When the polarity reversal detection is disabled, VGW-400 Series answer the call (establish call between VoIP and FXO) after time out to start Billing count purpose. After the DTMF digits dialing, VGW-400 Series send 18 3 with SDP to SIP Trunk to enable the voice path for VOIP side. This ring count is used for called ID detection and 2 stage dialing. z If the caller ID is sending between the first ring and
second ring, this parameter should be set to greater than or equal to 2.
z If the caller ID is sending before the first ring, this
Caller ID Mode
parameter can be set to greater or equal to 1. After the ring count was reached, VGW - 400 Series answer the call and play voice greeting file if 2-stage dialing is selected. Or, make the VOIP call out directly if hot line mode and number is selected. The detected Caller ID specification from the PSTN line based on selected country list or FSK or DTMF
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Chapter 10 SIP Trunk

The SIP trunk for VOIP outgoing call and incoming call can be co nfigured by administrator authority. There are up to 4 SIP trunk can be used.
Note: please don’t delete sip trunk, even it is unless at all, because it have to be used with Route plan.

10.1 Create SIP Trunk

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Item Explanation
VGW-400 Series
Trunk ID Register Type TEL No Proxy Server Proxy Server port Outbound Proxy Outbound Server Port:
SIP trunk ID 1 to 4 Register type is predefine or register The Tel no for the SIP account The SIP proxy server address The SIP proxy server port number The SIP outbound proxy server address The SIP outbound proxy server port
Item Explanation Trunk ID
SIP trunk ID 1-4
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Register Type
Domain Proxy Server Proxy Server Port Outbound Proxy Server Outbound Proxy server port Register Expires TEL No User ID User Password
Whether this account need register or not Register: When it is set to register, VGW-400 Series
sends REGISTER message to SIP proxy server for
registration. Predefine: When it is set to predefine, VGW-400 Series
DOES NOT send REGISTER message out. The SIP domain for register or call making SIP registrar server address SIP registrar server port number Outbound proxy server address Outbound proxy server port number
The default register expired for negotiation The registrar telephone number The SIP user ID for register and call making The SIP password for register and call making
Display Name Reject Anonymous Call Outgoing Caller ID
The SIP display name Reject the anonymous call The outgoing SIP caller ID mode.
-Display Name: The display name will be set according to the following type.
None: No display name will be used PSTN caller ID: The display name will be the collected PSTN
caller ID SIP display name: The display name will be the Display Name set in this SIP trunk. FXO T el NO: The di splay name will be the incoming FXO’s TEL No set on FXO lines.
-User ID: The SIP caller ID will be used according to the following type. z SIP user ID: If the SIP user ID is set, the SIP user ID set in this SIP trunk will be used and the domain/SIP proxy will be the host part. The SIP FROM header’s URL will be the
SIP_User_ID@Domain or SIP_User_ID@SIP_Proxy_Server.
z PSTN caller ID: If the PSTN caller ID will be used in SIP URL, the SIP FROM header’s URL will be
PSTN_Caller_ID@local_IP_address.
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z FXO Tel NO: If the FXO Tel NO will be used in SIP URL, the SIP FROM header’s URL will be
FXO_Tel_NO@local_IP_address.
The following guideline could be used for most cases:
1. If the VGW-400 Series in SIP proxy was handled as a gateway, please set both the display name and User Id to be “PSTN caller ID”.
2. If the VGW-400 Series in SIP proxy was handled as a subscriber , please set the display name to “PSTN caller ID” and user ID to “SIP User ID”.
For DNIS is Register TEL
Keep Alive Keep Alive Ti me (sec)
When you have a call from VoIP to FXO to call out to PSTN network, there are two methods can be used. ( FXO port dialing out only )
1-stage dialing: When there is an SIP trunk incomi ng call to
VGW-400 Series, it selects a free FXO port and dial out digits directly without doing DM and routes plan directly.
Note: If VGW-400 Series was configured to PABX Mode, the
incoming call from VoIP or FXO port only route to FXS port. However, the outgoing call from FXS port goes to either VoIP or FXO port depend on DM and routes plan.
2-stage dialing: When there is an SIP trunk incomi ng call to
VGW-400 Series, it answers this call and play dial tone to SIP trunk to wait for SIP trunk user to dial digits and send
these digits to FXO/PSTN network one by one. Enable or Disable it. Specify interval time to send SIP register message to proxy server.
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Chapter 11 Route Plan

The routing policy is the core feature of VGW-400 Series. The poli cy is based on incomi ng call type, destination, length and prefix code to determine the outgoing call routes and process. There are three routes to go for each incoming call port as follows.
The following rules do not apply to PABX Mode. (For VGW-402 only)
1. VOIP incoming call to VGW-400 Series, it routes to either FXO or FXS interface and vice versa.
2. FXO incoming call to VGW-400 Series, it routes to either VoIP or FXS interface and vice versa.
3. FXS incoming call (it means FXS off hook and dialing out) to VGW- 400 Series, it routes to either FXO or VoIP interface and vice versa.

11.1 For PABX Mode interface

For this application, FXS outgoing call was routed to either VOIP or FXO and vice versa. The default route is that VOIP incoming call was routed to FXS and FXS call was routed to VOIP network.
The PABX mode follows these rules to routes call as follows.
1. When FXO has incoming call to VGW-400 Series, it routes to FXS port only.
2. When VoIP has incoming call to VGW-400 Series, it routes to FXS port only.
3. When FXS make a dial out, the routes call was redirected to either VoIP or FXO according to this gateway’s DM and routing plan.
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Item Explanation
VGW-400 Series
Incoming Call Type: Matched Prefix Matched Incoming List Matched Length
Outgoing Type
Create Route Plan>
Click “Routes Plan” then create a new routing policy.
The incoming call port is FXS or VOIP. Matched DNIS (called number) prefix Matched DNIS incoming interface target Matched DNIS (called number) length. The zero (0) mean no limitation of length. The outgoing call from FXS port can only go to either FXO or VOIP.
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Item Explanation
VGW-400 Series
Incoming Call Type
Matched Prefix Matched Incoming List
Matched Length
No Answer Timeout
Create Route Plan>Primary Route
Item Explanation
Incoming call type
z VOIP: The incoming SIP call type z FXS: The FXS extensions incoming call type
Matched DNIS (called number) prefix Matched DNIS incoming interface target For FXS incoming call type, the incoming target will be the line ID. Only the call is coming from the selected line will be accepted for this route. Matched DNIS (called number) length. To ignore the length, please set to 0. How long does the hunting continue to next when the called target doesn’t answer.
Outgoing Type Hunting Type
DM Group
Create Route Plan>Backup Route
Item Explanation
Backup Route Active Outgoing Type
Outgoing call type (FXO or VOIP or FXS) The hunting method can be used for this route. z Priority Ring: The call was hunted based on the routing list order one by one. z Cyclic Ring: The call was hunted based on the cyclic basis. This is the recommended method.
z Routing List: The routing target list was used for this route.
Select DM group 1 to 4 in case that it requires a DM routes (for example, remove the prefix) before making the call.
Active the backup routes or not. Define backup routes outgoing call type.
Hunting Type
Routing List
The hunting method was used for this route. Please refer to the Primary Route. The backup routing target list was used for this route
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Route DM Group:
2 special default route, “VOIP Default Route” and “FXS default Route”, are used as the default routing when there is not any other routing are matched. It is not recommended to disable these 2 default routes. The FXS default route was used as FXS outgoing call’s default routes. VOIP default route is used as VOIP incoming call’s default routing.
In this mode all of the VOIP and FXO incoming calls were forced to route to FXS port. The VOIP incoming call can’t route to FXO port to dial out.
Select DM group 1 to 4 in case that the backup required the DM before making the call. The DNIS is unchanged by the primary route DM and the same as the DNIS before routing. For example, the DNIS is 886282265699 and primary DM group remove 886 and use it (DNIS = 282265699) to make call. When backup route is started, the DNIS is still unchanged as
886282265699. This makes the DM easy to predict and implement.

11.2 For Non-PABX Mode interface

For this interface, it could be routed to VOIP and FXO and FXS and vice versa. You can ignore the routing plan if you don’t need it for FXS interface.
Item Explanation
Incoming Call Type Matched Prefix Matched Incoming List Matched Length
Incoming call type (VOIP or FXS or FXO) Matched DNIS (called number) prefix Matched DNIS incoming interface target Matched DNIS (called number) length
Outgoing Type
The outgoing call type (FXS or VOIP or FXO)
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Item Explanation
Incoming Call Type
Matched Prefix Matched Incoming List:
Incoming call type
z VOIP: The incoming SIP call type z FXO: The PSTN incoming call type z FXS: The FXS outgoing call type
Matched DNIS (called number) prefix Matched DNIS incoming interface target z For VOIP incoming call type, the incoming target will be the
SIP trunk ID. Only the call from the selected SIP T runk will be accepted for this route.
z For PSTN (FXO port) incoming call type, the incoming target
will be the line ID. Only the call is coming from the selected line will be accepted for this route.
z For FXS incoming call type, the incoming target will be the
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line ID. Only the call is coming from the selected line will be accepted for this route.
Matched Length
No Answer Timeout
Create Route Plan>Primary Route
Item Explanation
Outgoing Type Hunting Type
Matched DNIS (called number) length. To ignore the length, please set to 0. How long does the hunting continue to next when the called target doesn’t answer.
Outgoing call type (FXO or FXS or VOIP) The hunting method was used for this route. z Priority Ring: The call was hunted based on the routing list
order one by one.
z Cyclic Ring: The call was hunted based on the cyclic basis.
This is the recommended method.
z Routing List: The routing target list was used for this route.
DM Group
Create Route Plan>Backup Route
Item Explanation
Backup Route Active Outgoing Type Hunting Type
Routing List Route DM Group
Select DM group 1 to 4 in case that it requires a DM (for example, remove the prefix) before making the call.
Active the backup route or not. The backup route outgoing call type. The hunting method will be used for this route. Please refer to the Primary Route. The backup routing target list will be used for this route. Select DM group 1 to 4 in case the backup required the DM before to make the call. The DNIS is unchanged by the primary route DM and same as the DNIS before routing. For example, the DNIS is 886282265699 and primary DM group remove 886 and use it (DNIS = 282265699) to make call. When backup route is started, the DNIS is still unchanged as 886282265699. This makes the DM easy to predict and implement.
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3 special default route, “VOIP Default Route” and “FXO default Route” and “FXS default Route” are used as the default routing when there is not any other routing are matched. It is not recommended to disable these 3 default route. The FXO defa ult route is used when a FXO incoming call’s default routing. VOIP default route is used for a VOIP incoming call’s default routing. FXS default route is used when a FXS outgoing call default was routing.
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Chapter 12 Status

12.1 Device Status

VGW-400 Series
Item Explanation
Model MAC-Address Network Type IP-Address IPV6 IP-address: Firmware
The model number The MAC address of VGW-400 Series The Network Interface Type Settings IP address is using Display IPV6 address The firmware version and released information

12.2 Line Status

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Item Explanation
VGW-400 Series
Line Call State: Refresh Interval (second):
L1 to L4 The Line status for this line The time to refresh the status

12.3 SIP Trunk Status

Item Explanation
Account Registered Concurrent Call Refresh Interval
SIP trunk account The SIP trunk register status The concurrent calls are used for this SIP trunk The time to refresh the status
(second)
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Chapter 13 Maintenance

VGW-400 Series can be managed by this management page to upgrade firmware or Reset this device.
Item Explanation
Backup Restore Reset to Default Quick-Reset Reboot
Backup the system settings for restoring purpose Restoring the backup setting to this device Reset system setting to factory default value. Warm Reset without reboot this device. Reboot this device

13.1 Firmware Update

This maintenance page provides the firmware upgrade features.
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Appendix A - Default Setting

VGW-400 Series
Default WAN IP Default subnet mask Default Gateway Default PC IP Default Login User Name Default Login Password
172.16.0.1
255.255.255.0
172.16.0.254
192.168.0.1 admin admin
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Appendix B - Changing IP Address or forgotten admin password

To reset the IP address to the default IP address “192.168.0.1” (LAN) or reset the login password to default value, press the reset button on the front panel for more than 5 seconds After the device is rebooted, you can login the management Web interface within the same subnet of 192.168.0.xx.
Reset Button
After pressing the “Reset” button, all the system data will be reset to default; if possible, back up the config file before resetting.
.
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