PLANET VGW-400FO User Manual

4-Port SIP Internet Telephony Gateway

VGW-402 / VGW-400FS / VGW-400FO

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Copyright© 2014 by PLANET Technolog y 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 in 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 radi o interference, in whi ch case the user may be required to take adequate measures.

Federal Communication Commission Interference Statement

This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Part 15 of FCC Rules. These limits are designed to provide reasonable protection against

4-Port SIP Internet Telephony Gateway

VGW-400 Series

harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

1. Reorient or relocate the receiving antenna.

2. Increase the separation between the equipment and receiver.

3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

4. Consult the dealer or an experienced radio technician for help.

FCC Caution:

To assure continued compliance, for 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 authority to operate the equipment. This device complies with Part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

R&TTE Compliance Statement

This equipment complies with all the requirements of DIRECTIVE 1999/5/EC OF THE EUROPEAN PARLIAMENT AND THE COUNCIL OF 9 March 1999 on radio equipment and telecommunication terminal Equipment, and the mutual recognition of their conformity (R&TTE). The R&TTE Directive repeals and replaces in the directive 98/13/EEC (Telecommunications Terminal Equipment and Satellite Earth Station Equipment) as of April 8, 2000.

WEEE Caution

To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment, end users of

electrical and electronic equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 Planet Products, 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: 1.1 (

January, 2014)

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 Web 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).....................................................................................................34

4.4 URL Filter.......................................................................................................................................38

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 T one.................................................................................................................................................42

5.4 NAT Traversal.................................................................................................................................43

Chapter 6 VoIP Advance.............................................................................................................44

4-Port SIP Internet Telephony Gateway

VGW-400 Series

6.1 SIP...................................................................................................................................................44

6.2 SIP...................................................................................................................................................47

6.3 Ring.................................................................................................................................................49

Chapter 7 Dialing Plan ...............................................................................................................50

7.1 General............................................................................................................................................50

7.2 Dialing Rule ....................................................................................................................................50

7.3 Digit Manipulation..........................................................................................................................52

7.4 Phone Book.....................................................................................................................................53

Chapter 8 FXS Setting................................................................................................................54

8.1 FXS Line.........................................................................................................................................54

8.2 SIP Proxy.........................................................................................................................................57

8.3 Caller ID..........................................................................................................................................58

8.4 Others..............................................................................................................................................59

Chapter 9 FXO Setting ...............................................................................................................60

9.1 FXO line..........................................................................................................................................60

Chapter 10 SIP Trunk..................................................................................................................63

10.1 Create SIP Trunk...........................................................................................................................63

Chapter 11 Route Plan ...............................................................................................................67

11.1 For PABX Mode Interface.............................................................................................................67

11.2 For Non-PABX Mode Interface.....................................................................................................72

Chapter 12 Status.......................................................................................................................76

12.1 Device Status.................................................................................................................................76

12.2 Line Status.....................................................................................................................................76

12.3 SIP Trunk Status............................................................................................................................77

Chapter 13 Maintenance.............................................................................................................78

13.1 Firmware Update...........................................................................................................................78

Appendix A – Default Setting.....................................................................................................79

Appendix B - Changing IP Address or Forgotten Admin Password ......................................80

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 enterprise-class 4-port SIP VoIP Gateway series. 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-dist ance 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 help f ul 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.

4-Port SIP Internet Telephony Gateway

Enhanced, Full-Featured Business Gateway

VGW-400 Series

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 VoIP call over Internet or VPN network. This device is suit able 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.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Secure, High-Quality VoIP Communication

It can effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS (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 functio n, help user 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).

4-Port SIP Internet Telephony Gateway

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)

VGW-400 Series

 IPv6 only, IPv4 only or dual stack  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

4-Port SIP Internet Telephony Gateway

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

4-Port SIP Internet Telephony Gateway

VGW-400 Series

1.2 Package Contents

Thank you for purchasing PLANET Internet Telephony Gateway system, the 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 the above items are damaged or missing, please contact your dealer immediately.

1.3 Physical Specifications

¾ Dimensions

Dimensions Weight

175 x 32 x126 mm 550g

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)

4-Port SIP Internet Telephony Gateway

Rear Panel of the VGW-400 Series (VGW-400FO)

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 will 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

FXO Ports

Press and hold over 5 seconds to reload factory default setting,

which will erase all existing settings configured on this gateway.

The status LED for FXS port, will light up amber orange when the

connected phone’s handset is lifted, or when the connected phone is

engaged in a conversation. It will flash amber orange when there

is an incoming call.

The status LED for FXO port will remind you that there is no PSTN

line connected. When PSTN line is connected and there is no

talking, the LED is OFF. When a line is using, the LED becomes

steadily light up.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

LAN

WAN

DC 12V

10/100Base-TX RJ-45 socket for LAN port connects to PC for

management purposes.

10/100Base-TX RJ-45 socket for WAN port connects to wide area

network.

The power socket, input AC 100V~240V; output DC12V, 1.5A

Button Action Description

Reset

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.

1.4 Specifications

Product

Hardware

WAN

LAN

Voice

Protocols and Standard

Data Networking

VGW-400 Series

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)

(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

Vendor Class ID

IP / ICMP / ARP / RARP / SNTP

Static IP

DHCP Client (RFC 2131), WAN port

DHCP Server, LAN port

NAT Server (RFC 1631)

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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

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

Voice Gateway

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)

REFER (RFC3515)

Supports Outbound Proxy

Supports Primary and Backup SIP Server

Supports STUN NAT Traversal

Supports “rport” parameter (RFC 3581)

Configure SIP local Port

Audio Codec

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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

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

Functions

Call Functions

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

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 supports STUN, UPNP and Behind NAT

Out-Band DTMF with RFC2833 and SIP Info

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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

FXO/FXS Line Configuration

Flexible Routing Plan

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

Outgoing SIP Caller ID Selection

Supports 4 SIP Trunk Accepts desired SIP Proxy incoming calls Only

Prefix Match and Length

Priority Ring

Cyclic Ring

Simultaneous Ring

Programmable Hunting Cycle

Backup Routes with Digit Manipulation

Default Routes

Flexible Dial Plans

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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: ranging from 80 to 800 ms

On-Hook Voltage -48Vdc

FXS Analog 2-wire Interface

FXO Analog 2-wire Interface

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 Languages (Administrator, Supervisor and User)

HTTP/HTTPS Service Access limitation from WAN port

Management

Configure Service ports at HTTP, HTTPS and telnet Services

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

Environments

4-Port SIP Internet Telephony Gateway

VGW-400 Series

SNMP V2 / Trap

Configuration Backup/Restore

Dual Firmware Image Backup

Reset to Factory Default

Power Requirements

Operating Temperature

Operating Humidity

Weight

Dimensions (W x D x H)

Emission

Connectors

12V DC, 1.5 A

0 ~ 45 degrees C

10%~90% relative humidity, non-condensing

550g

175×32×126 mm

CE, FCC, RoHS

Two 10/100Base-TX RJ-45 Ethernet ports

Four RJ-11 ports

DC power jack

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 LAN 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

For security reason, please change and memorize the new password after this first setup.

4-Port SIP Internet Telephony Gateway

2.2 Configuring the Network Setting

Step 1. Go to Device Setting → Network

Network setting page

VGW-400 Series

Step 2. Edit your WAN port IP information.

There are three types of IP Support -- IPV4 Only, IPV4 / IPV6, IPV6 Only. There are also three types of WAN port connection -- Static IP, PPPoE (Point-to-Point Protocol over Ethernet) and DHCP. You can find detailed setting process in the user manual.

Selection of IP Support / Network Connection Type

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 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.

. After the device is

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 3 Device Setting

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

Network Configuration

3.1 Network Configuration

Figure 2-1 network setting

Parameter Description: Setting

WAN Setting:

z IP Support: IP st ack to be supported (IPV6 and IPV4 or IPV6 or IPV4 only)

1 Network Type

2 IP Address

3 Net mask

4 Default Gateway

DHCP Tag (60 is

optional)

DHCP Tag (61 is

optional)

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

4-Port SIP Internet Telephony Gateway

VGW-400 Series

8 IPV6 IP Address

9 IPV6 IP Gateway

10 IPV6 IP Prefix Length

11 DNS Server 1

12 DNS Server 2

13 VoIP VLAN

14 VoIP VLAN ID (2-4096)

LAN Setting:

1 Management Mode

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 WAN port can be only accessed by VLAN. If it is required to manage the VGW Gateway series, administrator can use LAN port to access this gateway instead.

VLAN ID range to be used

This LAN port is used for management purposes, not used for register to SIP Server or data/voice routing.

2 NAT Mode

3 IP Address

4 Net Mask

5 Bridge Mode

DNS Setting:

1 DDNS

2 Domain Name

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 the command of “NAT setting” on the left side for how to define DHCP IP address.)

IPV4 address IPV4 network subnet mask

In this mode, both WAN and LAN ports are configured to Switch/Hub features. LAN port has access to WAN port directly.

It supports Planet DDNS, Easy DDNS and DynDNS or disables the DDNS feature.

Input your domain name

3 User Name

4 Password

Input your user name Input your password

4-Port SIP Internet Telephony Gateway

For more detailed information on Planet DDNS function, please refer to the Appendix: Planet DDNS page.

3.2 Device Time Setting

The VGW-400 series supports SNTP with time zone and daylight saving.

Device Setting > Time

VGW-400 Series

Configure Time Setting

Parameter Description:

1 Current Time

2 NTP Time Server

NTP Refresh

Interval(sec)

4 Time Zone

5 Daylight Saving

6 Daylight Bias

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

- Customized: Use a user defined time zone

Auto adjust daylight saving time 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

7 Daylight Start

- Month: 01 to 12

- Week Day: Sunday to Saturday

- Apply Week (Day:01 to 05, Specifies the occurrence of

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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

8 Standard Start

- 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

4-Port SIP Internet Telephony Gateway

3.3 Device Advance Setting

Parameter Description:

HTTP Service

HTTPS Service The https web service port (the default is 443)

Telnet Service The telnet service port (the default is 23)

The Administrator Web service port (the default is 80)

VGW-400 Series

When clicking the disable option, the Web service will be

HTTP/HTTPS Service

Access on WAN

rejected on WAN port. So, please be careful with this function. If

you want 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 levels of users can be used, administrator, supervisor and user. Each level of users has a

different predefined access level.

Extension Settings

Item Explanation

Administrator

Supervisor

The administrator level user who has full access authority to VGWGateway series. The supervisor level user who has limited administrative access right.

User

features.

The user access right which only allows setting some user related

User ID Password Confirm

Confirm new password again

Password Language

The desired web page language used when the account is login.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

3.5 Debug Setting

The 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 when necessary and under Planet FAE’s instruction.

Item Explanation

Syslog Check for Start

Enable or disable to send system information to syslog server or not

Always send syslog or only during a specified time range.

anytime Syslog Start (YYYY/MM/DD HH:MM) Syslog Stop (YYYY / MM / DD HH:MM) Syslog Server Syslog Port DSP Debug DSP Capture

Always send syslog or only during a specified time range.

The syslog stop sending time

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

Server DSP Capture Port

syslog capture server service port (default is 50000)

4-Port SIP Internet Telephony Gateway

VGW-400 Series

3.6 Event Notice

VGW Gateway series can send Syslog Event Notice when it has the following cases:

1. Register Failure or re-registered

2. FXO RJ-11 cable is plugged or unplugged

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.

4-Port SIP Internet Telephony Gateway

9510: (This feature is not yet available now. Please don’t select at present.

Select HTTP:

Item Explanation

VGW-400 Series

Http Config URL Refresh interval(minute) User ID Password

3.8 SNMP

internal use only interval to check whether there is a new configuration/firmware

or not in minutes specify the Login ID for http authentication

specify the password for http authentication

SNMP Agent:

Item Explanation

SNMP Agent Enable SNMP or not Read Only

The community name to read through SNMP protocol

Community Name Read Write

The community name to read and write through SNMP protocol

Community Name SNMP Agent Access

Enable SNMP to be accessed through WAN port or not

on WAN

Trusted Peer:

Item Explanation

Type

IP address

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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. The IP address for a trusted peer

Subnet Mask

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 being used for the VGW-400 series to become an inter-connection in between PSTN Lines and analog trunk lines from the traditional PABX.

When this mode is changed (enables to disable or disable to enable), it will clear all of the route plans and return to the default route.

The 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 corre sponding FXS directly (Line1 to Tel1, Line2 to Tel2) (For VGW-402 Only)

2. For FXS outgoing call, it will be routed to VoIP except the prefix set in FXO dialing prefix.

3. For VoIP incoming call from SIP Trunk number, it will be routed to FXS based on the called number.

If you are dialing to SIP trunk number, and hear the dial tone from the VGW Gateway series, please check the SIP Trunk configuration. It might be configured to option mode at “1 stage dialing”.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

4. When VoIP call fails to be called out su ch as registe r failure (this mean s registrat ion to proxy accounts failed, but not the SIP Trunk number) or network issue. The call will be routed to FXO as a backup.

5. When the VGW-400 series is malfunctioned, IP netwo rk di sconnecti on o r powe r fail. All calls will be directly bypassed to FXO automatically.

Item Explanation

PABX Mode Enable or Disable PABX mode, default is “Enable”

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 4 NAT Setting

The 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 IP Client Range End IP Default Gateway Submask DNS Server 1 DNS Server 2

Enable DHCP server or not

Specify DHCP client lease start IP

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

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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: tot

Item Explanation Priority 1 Priority 2 Priority 3

al bandwidth

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

4-Port SIP Internet Telephony Gateway

Specifics

Item Explanation

VGW-400 Series

Priority 1 – Download Priority 2 – Download Priority 3 – Download Priority 1 – Upload Priority 2 – Upload Priority 3 – Upload

highest priority download bandwidth normal priority download bandwidth low priority download bandwidth highest priority upload bandwidth normal priority upload bandwidth low priority upload bandwidth

In order to set which target belongs to which priority, the following are the setting methods for target’s priority.

IP Target

4-Port SIP Internet Telephony Gateway

Item Explanation

Priority Priority value for the target Type The target type is set to IP

Unique IP or a range of IP addresses

¾ Unique:

Configure Type

Port Target

 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

VGW-400 Series

Item Explanation

Priority Priority value for the target Type 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

Configure Type

 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

Application Target

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Item Explanation

Priority Type Application

DSCP target

Priority value for the target Application The list for the application

Item Explanation

Priority

Priority value for the target

Type DSCP

The VGW-400 series supports the firewall features below.

DSCP value The DSCP will be mapped to the priority

4.4 URL Filter

Item Explanation

4-Port SIP Internet Telephony Gateway

VGW-400 Series

URL Filter

The specified URL will be blocked

4.5 IP Filter

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

For the MAC address to be blocked, please follow these formats.

4.7 APP Filter

Item Explanation APP Filter

Application to be blocked

4-Port SIP Internet Telephony Gateway

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 forwarded. If you are using only 1

port, please set the starting equal to stopping port TCP, UDP or both are used for port block

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

5.1 SIP

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 5 VoIP Setting

Item Explanation Session Timer Session Expiry (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)

The SIP local service port (default is 8080) Quality of Service Type for SIP signaling

Only accept the call coming from the SIP proxy. Does not accept peer to peer call in this mode

- None: Disable PRACK

- Supported: When selecting this mode, 100rel will be added

to the support list. It indicates the VGW-400 series can support the PRACK but not mandatory.

- Require: PRACK is mandatory required.

- None: Not using QOS Tag and not enables QOS.

- DiffServ: Differentiated Services Value. Input DSCP value

0-63 for DSCP

- TOS: Type of Service which include IP precedence value and TOS.

4-Port SIP Internet Telephony Gateway

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

The preference codec priority G.711 u-Law payload size G.711 A-law payload size G.729A payload 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 Remote 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

DTMF Relay

Silence Suppression

SIP INFO for DTMF relay. SIP INFO: Use SIP-INFO DTMF relay

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 detecting silence for 60 seconds. It will hang up the call (For

FXO use)

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Disable: Send silence packets as normal voice packet (no silence detection) The RTP starting port. Each channel will be added additional

RTP Basic Port

RTP QoS Type

10. For example, the RTP basic port is 16384 and thus call 1 will use 16384 while call 2 will use 16394, etc. 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 set up 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 receives the

“bye” from IP side or waits time out by analog phone which keeps picking up the handset, it will send busy tone to analog phone). To recognize the correct disconnect tone is very important for

PSTN status supervision to release FXO port after call is 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 which are not shown in the list, please select a closed country and edit tone parameters to match your country. You can send an email with the tone definition to Planet if you would like to put your country tone in the list.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

5.4 NAT Traversal

The VGW-400 series supports the following NAT traversal methods when it is placed behind the router .

NAT Traversal:

Item Explanation 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 detects the current NAT is type 3, it stops the STUN feature operation.  STUN Server: STUN Server IP address No matter which NA T type server is u sed, STUN is always 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

6.1 SIP

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 6 VoIP Advance

Item Explanation

SIP on hold message sending method. Send Only: Set the SDP media to send only when sending an on-hold SIP message.

SIP Hold Type

SIP Compact Form

0.0.0.0: Set the SDP connection to 0.0.0.0 when sending an

on-hold SIP message. Inactive: Set the SDP media to inactive when sending an on-hold SIP message. Enable SIP compact form or not. When enabling this feature,

4-Port SIP Internet Telephony Gateway

VGW-400 Series

the connected SIP proxy is required to support compact form. Who will send dialog to keep message alive (re-invite or

Session Refresher

SIP T1 (msec)

SIP T2 (msec)

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) ** T4 represents the amount of time the network takes to clea r

SIP T4 (msec)

Invite Linger Timer

General Linger Timer

Cancel General No Response Time (msec)

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, the User Agent waits for cancel General No Response Timer milliseconds before timeout termination if there is no response

4-Port SIP Internet Telephony Gateway

VGW-400 Series

for the cancelled transaction(Default Value: 10,000 ms).** After sending a General request, the User Agent waits for a

General Request Timeout Timer (msec)

Cancel Invite No Response Timer (msec)

Provisional Timer (msec)

First Response Timer (msec)

final response general Request Timeout Timer millisecon ds 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. You can Enable or Disable the MWI subscription. The default is

MWI Subscription Expiry (sec)

Line Congestion Code

SIP-Info Flash Mode

Encrypt

600 sec. If a new voice mail arrives, 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

4-Port SIP Internet Telephony Gateway

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 / ATA / 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 Time (msec)

96 or 101. It is recommended to use 101. When generating DTMF, the DTMF ON time will be sent (default value is 70 ms) When generating DTMF, the DTMF OFF time will be sent (default value is 70 ms) The minimum DTMF ON time period will be processed as a regular DTMF event. A smaller ON time less than this will be

4-Port SIP Internet Telephony Gateway

VGW-400 Series

ignored. The default value is 60ms. The minimum DTMF OFF time for the same DTMF value. A

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)

Jitter Opt. Factor

Impedance

smaller OFF time less than this and the new DTMF digit is the same as previous one will be handled as 1 digit only (the same digit but 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 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)

4-Port SIP Internet Telephony Gateway

6.3 Ring

The ring cadence, voltage and frequency were configured to the phone.

Item Explanation

VGW-400 Series

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

4-Port SIP Internet Telephony Gateway

7.1 General

Item Explanation

VGW-400 Series

Chapter 7 Dialing Plan

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.

4-Port SIP Internet Telephony Gateway

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:

The prefix to be matched The digits will be received based on the Dialed Prefix.

Prefix Max. Digits

09 10 0 10 00 15 1 3 2 8 3 8 4 8 5 8 6 8 7 8 8 8 9 8

4-Port SIP Internet Telephony Gateway

VGW-400 Series

7.3 Digit Manipulation

The Digit Manipulation (DM) will be processed based on prefix and DM group after the DNIS (Called Party) is determined.

Item Explanation

Different DM groups have dif f erent applications as follows.

 FXO: This DM group is used for FXO port with 2-stage

dialing. Af ter the DNIS (Called party messages) is

DM Group

Matched Prefix

Matched Length

Start POS Stop POS

collected, this DM group will be processed before entering the routing procedure.

 FXS: This DM group is used for FXS dial out.  VOIP: This DM group is used for VOIP incoming call. Af ter

the DNIS is collected in 2-stage dialing or 1-stage 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 The start digit position to be replaced The stop digit position to be replaced

Replace Value

The value to be replaced

4-Port SIP Internet Telephony Gateway

Example of Digit Manipulation Settings:

VGW-400 Series

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

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.

Item Explanation

Name

Tel No

This field supports called number only. If you enter words or text here, it will route 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

Export Import

5060) To back up the phone book records To reload setting of phone book

4-Port SIP Internet Telephony Gateway

Chapter 8 FXS Setting

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

8.1 FXS Line

Item Explanation

VGW-400 Series

Line ID State Tel. No Hotline T el.

FXS line The line is active or not The telephone number of each FXS port If hot line is enabled, this field shows the hot line number

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Item Explanation Line ID

Line Type FXS or FXO (depending on device model).

Line State

Forward reasons:

FXS Line number (T1 to T2)

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 is not

answered after any answer timeout.

 Forward Tel.: The telephone number will be forwarded once

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Forward mode is activated.

No Answer Timeout (seconds)

Call Waiting

Reject Anonymous Call Hot Line

Hotline T el

Polarity Reversal Generation

Current Drop Generation

The no answer timeout will be used (default is 120 sec)

Enable call waiting or not. When call waiting mode is disabled, the second incoming call will be rejected. Reject the anonymous incoming call or not Enable to disable hot line feature The number will dial automatically after the user picks up the phone. Enable Polarity Reversal of tip/ring of RJ-11 phone line for FXS as billing signal or not. When an 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 reverses back as a billing stop signal. Enable current drop (0 voltage) when VoIP is disconnected (Remote party drops the call).

Input (Encode) Gain Output(Decode)Gain

Fax Relay

Voice Mail Subscription

Caller ID Mode

SIP Caller ID Mode

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 p hone.  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 registered message out.

 Internal: When it is set to internal, VGW-400 Series does

not send registered message out. The FXS line still can use SIP trunk for VoIP call or call locally.

Tel No User ID User Password Display Name

The registrar telephone number The SIP user ID for register and call making The SIP password for register and call making The SIP display name

4-Port SIP Internet Telephony Gateway

8.2 SIP Proxy

The SIP proxy server defined here is dedicated to FXS lines.

VGW-400 Series

Item Explanation Domain Primary Proxy Server Primary Proxy Server Port Outbound Proxy Server Outbound Proxy Server Port Primary Proxy Server Keeps Alive Keep Alive Time(sec)

Secondary Proxy

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 NAT to keep the port alive

Specify time to send SIP registered message to proxy server. Enable secondary proxy or not. When enabling it, the primary and secondary proxies will be registered at the same time.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Secondary Proxy Server Secondary Proxy Port Secondary Outbound Proxy Server Outbound Proxy Server Port Register Expiry: Secondary Proxy server keep Alive Keep Alive Time(sec)

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 NAT to keep the port alive

Specify time to send SIP register message to proxy server.

8.3 Caller ID

The call ID sends to FXS port of the analog phone set to display caller name or phone number.

4-Port SIP Internet Telephony Gateway

Item Explanation

VGW-400 Series

Caller ID Mode

Polarity Reverse before Caller ID Dual Tone before Caller ID

Caller ID present

DTMF Caller ID Start Digit DTMF Caller ID Stop Digit

8.4 Others

Caller ID mode to be used for phone (FSK Bellcore, FSK ETSI, DTMF)

Start polarity reverse to FXS port before sending the caller ID

Send Dual Tone before caller ID (for FSK ETSI use only)

The timing to send the caller ID (Before the first ring, after the first ring, after the 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 #)

Flash time and current drop generation/detection time

4-Port SIP Internet Telephony Gateway

The FXO setting contains the FXO related parameters.

Item Explanation

VGW-400 Series

Chapter 9 FXO Setting

Line ID State Tel No Hotline T el

9.1 FXO line

FXO line The line is active or not The reference telephone number (e.g. PSTN Tel line) If hot line is set, this field shows the hotline number

4-Port SIP Internet Telephony Gateway

Item Explanation

VGW-400 Series

User ID User T ype Line State

Tel No

Polarity Reversal Detection

Current Drop for Disconnection

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 enabling the Polarity Reversal Detection feature, VGW-400 Series uses the polarity reversal signal once call is established for FXO outgoing call and start to count talking time for billing purpose. When disabling the polarity Reversal Detection, VGW-400 Series uses “Dialing Answer Delay

Time” command to set time (seconds) to start billing time once

SIP call is established. Use Line current drop as a disconnecting supervision to release FXO port. When remote PSTN side user drops call, the local PSTN switch sends Current drop signal to FXO port to

Incoming Call Handling

Playback Voice File

Repeat Count

Voice File Name (MuLaw-mono 8K)

recognize this situation. The call handling policy for an FXO incoming call.  Hotline Tel: When a PSTN Line incoming call is detected

and after the FXO answers this call based on the Ring Count Configuration, the VGW-400 series sends SIP call to the specified hotline tel number through the Route Plan.

 2 Stage Dialing: When a PSTN Line incoming call is

detected and after the FXO answers this call based on the Ring Count Configuration, VGW-400 Series answers this call and plays either Dial Tone or Voice Greeting file to PSTN side. And wait for th e 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 that the file format needs to be G.711U, 8K, 8 bits raw file.

(Used for FXO port Only)

Flash Time

Flash Time wi ll be sent to PSTN line.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Fax Relay

Input (Encode) Gain Output (Decode) Gain

Dialing Answer Delay Time (sec)

PSTN Answer Ring Count

To 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 answers the call (establish call between VoIP and FXO) after time out to start billing count purpose. After the DTMF digits dialing, VGW-400 Series sends 183 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 sent between the first ring and the

second ring, this parameter should be set to greater than or equal to 2.

z If the caller ID is sent before the first ring, this parameter

can be set to greater or equal to 1.

After the ring count is reached, VGW-400 Series answers the

Caller ID Mode

call and plays voice greeting file if 2-stage dialing is selected. Or, make the VOIP call out directly if hotline mode and number is selected. The detected Caller ID specification from the PSTN line based on selected country list or FSK or DTMF.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 10 SIP Trunk

The SIP trunk for VoIP outgoing call and incoming call can be configured by administrator authority . There are up to 4 SIP trunks that can be used.

Please don’t delete SIP trunk, even it is useless because it has to be used with route plan.

10.1 Create SIP Trunk

4-Port SIP Internet Telephony Gateway

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 predefined or registered 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

4-Port SIP Internet Telephony Gateway

Item Explanation

VGW-400 Series

Trunk ID

Domain Proxy Server Proxy Server Port Outbound Proxy Server Outbound Proxy Server Port Register Expiry Tel No User ID

SIP trunk ID 1-4 Whether this account needs to 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 REGISTERED 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

User Password Display Name Reject Anonymous Call

Outgoing Caller ID

The SIP password for register and call making The SIP display name Reject the anonymous call The outgoing SIP caller ID mode.

-Display Name: The display name will be set as follows:

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 Tel No: The display name will be the incoming FXO’s Tel No. set on FXO lines. User ID: The SIP caller ID will be used as follows: z SIP user ID: If the SIP user ID is set, the SIP user I D 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

4-Port SIP Internet Telephony Gateway

VGW-400 Series

PSTN_Caller_ID@local_IP_address.

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 guidelines could be used for most cases:

1. If the VGW-400 series in SIP proxy is handled as a gateway, please set both the display name and User ID to “PSTN caller ID”.

2. If the VGW-400 series in SIP proxy is handled as a subscriber , please set the display name to “PSTN caller ID” and User ID to “SIP User ID”. When you have a call from VoIP to FXO to call out to PSTN network, there are two methods that can be used. ( FXO port

For DNIS is Registered Tel

Keep Alive

dialing out only )

 1-stage dialing: When there is an SIP trunk incoming call to

the VGW-400 series, it selects a free FXO port and dial-out digits directly without doing DM and route plan directly.



If the VGW-400 series is configured to PABX mode, the incoming call from VoIP or FXO port only routes to FXS port. However, the outgoing call from FXS port goes to either VoIP or FXO port depending on DM and route plan.

 2-stage dialing: When there is an SIP trunk incoming call to

the VGW-400 series, it answers this call and plays 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.

Keep Alive Ti me (sec)

Specify interval time to send SIP registered message to proxy server.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 11 Route Plan

The routing policy is the core feature of the VGW-400 series. The p olicy is based on incoming 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 shown below.

The following rules do not apply to PABX mode. (For VGW-402 only)

1. VoIP incoming call to the VGW-400 series -- It routes to either FXO or FXS interface and vice versa.

2. FXO incoming call to the 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 the 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 is routed to either VoIP or FXO and vice versa. The default route is that VoIP incoming call is routed to FXS and FXS call is routed to VOIP network.

The PABX mode follows these rules to route call as follows:

1. When FXO has an incoming call to the VGW-400 series, it routes to FXS port only.

2. When VoIP has an incoming call to the VGW-400 series, it routes to FXS port only.

3. When FXS makes a dial out, the route call is redirected to either VoIP or FXO according to this gateway’s DM and routing plan.

4-Port SIP Internet Telephony Gateway

Item Explanation

VGW-400 Series

Incoming Call Type Matched Prefix Matched Incoming List

Matched Length

Outgoing Type

Create Route Plan>

Click “Route Plan” and 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) means no limitation of length. The outgoing call from FXS port can only go to either FXO or VoIP.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

4-Port SIP Internet Telephony Gateway

Item Explanation

Incoming call type

VGW-400 Series

Incoming Call Type

Matched Prefix

Matched Incoming List

Matched Length

No Answer Timeout

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 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?

4-Port SIP Internet Telephony Gateway

Create Route Plan>Primary Route

Item Explanation

VGW-400 Series

Outgoing Type

Hunting Type

DM Group

Create Route Plan>Backup Route

Item Explanation Backup Route 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 is used for this route.

Select DM group 1 to 4 in case it requires a DM route (for example, remove the prefix) before making the call.

Activate the backup route or not. Define backup route outgoing call type.

Hunting Type

Routing List

Route DM Group:

2 special default routes, “VoIP Default Route” and “FXS Default Route”, are used as the default routing when there is no other matched routing. It is not recommended to disable these 2 default routes. The FXS default route is used as FXS outgoing call’s default route. VoIP default route is used as VoIP incoming call’s default routing.

The hunting method is used for this route. Please refer to the Primary Route. The backup routing target list is used for this route Select DM group 1 to 4 in case the backup requires 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 removes 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.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

In this mode all of the V oIP and FXO incoming calls are forced to route to FXS port. The VoIP incoming call can’t route to FXO port to dial out.

11.2 For Non-PABX Mode Interface

For this interface, it could be routed to VoIP, 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 Outgoing Type

Incoming call type (VoIP or FXS or FXO) Matched DNIS (called number) prefix Matched DNIS incoming interface target Matched DNIS (called number) length The outgoing call type (FXS or VOIP or FXO)

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Item Explanation

Incoming call type

Incoming Call Type

Matched Prefix

Matched Incoming List

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 the VoIP incoming call type, the incoming target will be

z For the PSTN (FXO port) incoming call type, the incoming

z For the FXS incoming call type, the incoming target will be

the SIP trunk ID. Only the call from the selected SIP Trunk will be accepted for this route.

target will be the line ID. Only the call coming from the selected line will be accepted for this route.

the line ID. Only the call coming from the selected line will be accepted for this route

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 is used for this route. z Priority Ring: The call is hunted based on the routing list

order one by one.

z Cyclic Ring: The call is hunted based on the cyclic basis.

This is the recommended method.

z Routing List: The routing target list is used for this route.

DM Group

Create Route Plan>Backup Route

Item Explanation Backup Route Outgoing Type

Hunting Type

Routing List

Route DM Group

Select DM group 1 to 4 in case it requires a DM (for example, remove the prefix) before making the call

Activate 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 requires the DM before making 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 removes 886 and use it (DNIS = 282265699) to make a call. When backup route is started, the DNIS is still unchanged as 886282265699. This makes the DM easy to predict and implement.

Three special default routes, “VoIP Default Route”, “FXO default Route” and “FXS default Route”, are used as the default routing when there is no other matched routing. It is not recommended to disable

4-Port SIP Internet Telephony Gateway

VGW-400 Series

these three default routes. The FXO default route is used when there is an FXO incomi ng call’s default routing. VoIP default route is used for a VOIP incoming call’s default routing. FXS default route is used when an FXS outgoing call default is routing.

12.1 Device Status

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 12 Status

Item Explanation Model MAC Address Network Type IP Address IPV6 IP Address Firmware

The model number The MAC address of the VGW-400 series The Network Interface Type settings IP address is used Display IPV6 address The firmware version

12.2 Line Status

4-Port SIP Internet Telephony Gateway

Item Explanation

VGW-400 Series

Line Call Status Refresh Interval (second)

L1 to L4 The status of this line

The time to refresh the status

12.3 SIP Trunk Status

Item Explanation Account Registered Concurrent Call Refresh Interval (second)

SIP trunk account The SIP trunk register status The concurrent calls are used for this SIP trunk

The time to refresh the status

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Chapter 13 Maintenance

The 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

Back up the system settings for restoring purpose Restoring the backup setting to this device Reset system setting to factory default value. Warm reset without rebooting this device. Reboot this device

13.1 Firmware Update

This maintenance page provides the firmware upgrade features.

4-Port SIP Internet Telephony Gateway

VGW-400 Series

Appendix A – Default Setting

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

4-Port SIP Internet Telephony Gateway

VGW-400 Series

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 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.

. After the device is

PLANET VGW-400FO User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Chapter 1 Introduction

1.1 Features

1.2 Package Contents

1.3 Physical Specifications

1.4 Specifications

Chapter 2 Installation Procedure

2.1 Web Login

2.2 Configuring the Network Setting

2.3 Changing IP Address or Forgotten Admin Password

Chapter 3 Device Setting

3.1 Network Configuration

3.2 Device Time Setting

3.3 Device Advance Setting

3.4 User Login Setting

3.5 Debug Setting

3.6 Event Notice

3.7 Auto Provisioning

3.8 SNMP

3.9 PABX Mode

Chapter 4 NAT Setting

4.1 DHCP Srv. (DHCP Server)

4.2 UPNP (Universal Plug and Play Server)

4.3 Bandwidth (Bandwidth Control)

4.4 URL Filter

4.5 IP Filter

4.6 MAC Filter

4.7 APP Filter

4.8 Port Filter

4.9 Port Fwd

5.1 SIP

Chapter 5 VoIP Setting

5.2 Audio

5.3 Tone

5.4 NAT Traversal

6.1 SIP

Chapter 6 VoIP Advance

6.2 Audio

6.3 Ring

7.1 General

Chapter 7 Dialing Plan

7.2 Dialing Rule

7.3 Digit Manipulation

7.4 Phone Book

Chapter 8 FXS Setting

8.1 FXS Line

8.2 SIP Proxy

8.3 Caller ID

8.4 Others

Chapter 9 FXO Setting

9.1 FXO line

Chapter 10 SIP Trunk

10.1 Create SIP Trunk

Chapter 11 Route Plan

11.1 For PABX Mode Interface

12.1 Device Status

Chapter 12 Status

12.2 Line Status

12.3 SIP Trunk Status

Chapter 13 Maintenance

13.1 Firmware Update

Appendix A – Default Setting