NetComm V110 User Manual

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User Guide

V110, V210P, V211, V220

VoIP ATA

WELCOME TO THE WONDERFUL WORLD OF VOIP

Congratulations on your purchase of a NetComm VoIP ATA. Whether this is your ﬁrst VoIP device or you

are upgrading from another device we’re glad to have you as part of the NetComm family. The NetComm V110/V210P/V211/V220 ATA’s are true next generation technology and in addition to the fantastic savings you’re going to experience you’ll also get fantastic features such as routing, T.38 Fax and 3 way conferencing.

This Quick Start Guide contains everything you need to get you up and running for everyday use but if you need to delve into any of the more advance conﬁguration features available on your unit you’ll ﬁnd a comprehensive User Guide on the CD that accompanies the ATA.

Should you run into any problems with your unit (and let’s face it, we all need a little help from time to time) you can ﬁnd the answers to many questions in the support section of our website www.netcomm.com.au/Support/

So without any further delay it’s time to get you more familiar with your ATA so you can start using VoIP.

IMPORTANT NOTICE: Please ensure that the Broadband modem/router to be used with this ATA has been designed to handle VoIP trafﬁc. Most new broadband modem/router will be able to handle VoIP without any issues and do not need extra conﬁguration to handle VoIP service. However, old broadband modem/router might need to be conﬁgured correctly to support it.

Please check with the broadband modem/router manufacturers for more details.

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VOIP ATA FEATURES

Network Protocol Tone

• SIP v1 (RFC2543), v2(RFC3261) • Ring Tone

• IP/TCP/UDP/RTP/RTCP • Ring Back Tone

• IP/ICMP/ARP/RARP/SNTP • Dial Tone

• TFTP Client/DHCP • Busy Tone

• Client/ PPPoE Client • Programming Tone

• Telnet/HTTP Server

• DNS Client

• NAT/DHCP Server

Codec Phone Function

• G.711: 64k bit/s (PCM) • Volume Adjustment

• G.723.1: 6.3k / 5.3k bit/s • Speed dial key

• G.726: 16k / 24k / 32k / 40k bit/s (ADPCM) • Phone book

• G.729A: 8k bit/s (CS-ACELP) • Flash

• G.729B: adds VAD & CNG to G.729

Voice Quality Call Function

• VAD: Voice activity detection • Call Hold

• CNG: Comfort noise generator • Call Waiting

• LEC: Line echo canceller • Call Forward

• Packet Loss Compensation • Caller ID

• Adaptive Jitter Buffer • 3-way conference

IP Assignment Auto Provisioning

• Static IP • HTTP

• DHCP • FTP

• PPPoE • TFTP

DTMF Function NAT Traversal

• In-Band DTMF • STUN

• Out-of Band DTMF

• SIP Info

SIP Server Configuration

• Registrar Server • Web Browser

• Outbound Proxy • IVR/Keypad

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Firmware Upgrade Security

• TFTP • HTTP 1.1 basic/digest authentication for Web setup

• HTTP • MD5 for SIP authentication (RFC2069/ RFC 2617)

Interface Modem & Fax modes

1 WAN port interface

1 LAN port interface • T.38 support

1 PSTN port interface (FXO) (Optional)

1 VOIP port interface (FXS)

• G.711 fax/modem pass-through with fax/modem detection

PACKAGE CONTENTS

Your NetComm ATA package contains the following items:

• V2xx ATA

• Quick Installation Guide

• User Guide and Software on CD

• RJ-45 Straight-through Ethernet Cable

• DC12 volts/0.6 A Power Supply Unit

• One RJ-11 telephone cable (V210P / V211 only)

MINIMUM SYSTEM REQUIREMENTS

Before continuing with the installation of your Netcomm V2xx ATA, please conﬁrm that you comply with the minimum system requirements:

• Broadband (ADSL or Cable ) connection

• PC with TCP/IP networking protocol installed

• Windows XP/Vista PC for Easy Setup Utility

• VoIP account with a VSP (VoIP Service Provider)

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GETTING TO KNOW YOUR NETCOMM V2XX ATA

It is recommended that you take a moment to acquaint yourself with the indicator lights, ports and default settings of the V2xx ATA prior to commencing with installation.

Ports and Buttons

Port / Button Description

Power

Power Switch

WAN Port

LAN Port

RESET

LINE Socket

PHONE Socket

Connect the supplied power adapter to the ATA. The power requirement is DC12 volts/0.6 A

Note: Using a power supply with a different voltage rating will damage this product

Power on/off the NetComm ATA.

Connect to Broadband devices, such as an ADSL or Cable modem.

Connect to Ethernet network devices, such as a PC, switch, or router. Depending on the connection, you may need a cross over cable or a straight through cable

The Reset button will set the NetComm ATA to its factory default setting and reset the unit. You may need to reset the NetComm ATA into its factory defaults if you loose the ability to enter the NetComm ATA via the web interface, or following a software upgrade.

Connect a telephone cable between the NetComm ATA line jack and a wall jack.

Connect a standard telephone handset to the NetComm ATA phone jack using a telephone cable.

V110 (WAN + 1 LAN + 1 FXS)

V220 (WAN + 1 LAN + 2 FXS)

V211 (WAN + 1 LAN + 1 FXO +1 FXS)

V210P (WAN + 1 LAN + 1 PSTN pass through +1 FXS)

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LED Indicators

LED Description

PWR

SIP

ETH

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This LED illuminates to indicate the system is powered.

This LED illuminates when the NetComm ATA is registered successfully to the SIP server.

This LED illuminates when a connection is established to WAN/LAN port and ﬂashes when WAN/LAN port is sending/receiving data.

Default Settings

LAN (Management)

Static IP Address:

Subnet Mask:

Default Gateway:

WAN (Internet)

WAN mode:

Modem Access

Username:

Password:

Many conﬁguration changes made using the web interface will only become active after clicking the Save & Reboot button on the Save Savings / Reboot page in the web conﬁguration page.

Note: Certain voice parameters do not require a Save & Reboot to take effect. These Voice Parameters will take effect on the next voice call after the Voice Parameter is entered and submitted. However, if Save & Reboot is not done, then these Voice Parameters will not be saved over a power cycle. The Voice Parameters that can be changed “on the ﬂy” are noted in the respective sections.

192.168.22.1

255.255.255.0

blank

DHCP

admin

Restoring Factory Defaults

The reset button will reset the ATA to its factory default conﬁguration. Occasions may present themselves where you need to restore the factory default settings on your ATA. Typical situations are:

• You have lost your password and unable to login to the ATA;

• You have purchased the router from someone else and need to reconﬁgure the device.

• You are asked to perform a factory reset by a member of the excellent NetComm Support Staff.

In order to restore your router to its factory default settings, please follow these steps:

• Ensure that the ATA is powered on (for at least 20 seconds).

• Use a paper clip or a pencil tip to depress the reset button for ten seconds and release. At this point, the

reset is in progress. Do not power off the unit at this point.

• After the ATA reboots, the default settings are now restored. This entire process takes several minutes to

complete.

• Once you have reset the ATA to its default settings you will be able to access the device’s web

conﬁguration using http://192.168.22.1 with username “admin” and password “admin”.

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HARDWARE INSTALLATION

1. Connect the WAN port to your broadband router using the Ethernet cable supplied. If you would like

your PC to be connected to the network through the V100/V210P/V211/V220, you can connect another Ethernet cable (not supplied) from your PC to the LAN Ethernet port. This enables your PC and the ATA to share one Ethernet port on your router/switch.

V211 and V210P

2. Connect your telephone handset to the Phone port. For V220, you can connect an additional handset to

Phone 2 port.

Notes: A second VoIP account is needed unless your VoIP provider supports multiple connections from the one account and both ports need to be conﬁgured individually (even when using one account that supports multiple connections).

3. For V210P or V211 unit, connect the Line port to your PSTN wall socket.

4. Connect the AC power adaptor and press the power button to turn the ATA on. The ATA will boot up and in

a few minutes, the Power and Ethernet Lights will light up.

Note: Only use the power adapter supplied with the NetComm ATA. A different adapter may damage the product.

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V110

V220

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SOFTWARE CONFIGURATION

Easy Setup Utility (for Windows Only)

1. For easy conﬁguration, insert the included CD into your CD-ROM drive. The CD should auto-start. If it

does not start, click on Start -> Run and type in D:\Easysetup\vbpES.exe (where D is the drive letter of your CD-ROM drive).

2. The default Internet access is DHCP Client Mode. Click on WAN Conﬁguration to change the internet

access type.

3. Enter the SIP Proxy Domain, Proxy Server, Outbound Proxy, Username, Auth. ID and Auth. Password

provided by your VSP.

4. And then click on Setup, the software will start to conﬁgure the NetComm ATA. Follow the instructions of

the Easy Setup utility which will guide you to complete the conﬁguration.

5. After a few minutes, the NetComm ATA should report that the setup has been successful.

Note: The easy setup utility can not be used to retrieve details. The utility will initialize the unit back to default settings when executed.

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Conﬁguration via a web browser

If you do not have a Windows PC available, or you want to perform more complex conﬁguration, you can conﬁgure the NetComm ATA using a web browser.

Accessing the web conﬁguration

1. Open your Web browser (e.g. Internet Explorer, Firefox).

2. Enter the LAN port default IP address of the NetComm ATA (e.g. http://192.168.22.1) in the address bar

and press Enter.

3. At the login screen, enter the gateway username and password. The default login username is admin,

and the default login password is admin.

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WAN Conﬁguration

There are three types of WAN connection that you can use

• Static IP, to specify the IP address manually. Mostly used when you connect the V2xx ATA to a broadband

router with out a DHCP server.

• Dynamic IP, to connect to a service which has a DHCP server on the network. Mostly used when using the

V2xx ATA with an Optus cable modem or connecting the V2xx ATA to another router / gateway.

• PPPoE Client, to connect to a network with PPPoE service. Mostly used to connect the V2xx ATA to a

bridge ADSL modem.

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Static IP Conﬁguration

1. Click on WAN on the left menu.

2. Change the IP Type to DHCP.

3. Enter the required information, IP Address, Subnet Mask, Gateway, DNS Server1, DNS Server2 and Host

Name in the respective ﬁelds.

4. After that, click the Submit button.

Option Description

LAN Mode

IP Type

Mask

Gateway

DNS Server 1

DNS Server 2

MAC

Host Name

Conﬁgures the VoIP Gateway as a router or bridge.

Fixed IP address allows the user to manually assign an IP address. DHCP mode allows the VoIP Gateway to automatically obtain an IP address from a DHCP enabled server on your network. PPPoE allows the VoIP Gateway to use a PPPoE connection. In this mode, enter the PPPoE Username and Password in the ﬁelds under PPPoE settings.

The unique address of the VoIP Gateway on your network.

The subnet mask of your network.

The default gateway address through which the VoIP Gateway communicates with the Internet. This is usually your modem / router.

The Domain Name Service that the NetComm ATA uses to resolve domain names to IP addresses. This is usually your modem / Router or the primary DNS server speciﬁed by your Internet Service Provider (ISP).

The Domain Name Service that the NetComm ATA uses to resolve domain names to IP addresses. This is usually your modem / router. DNS 2 can be a different address to DNS 1 to provide a backup DNS server if DNS 1 fails.

The Ethernet MAC address of the WAN interface.

The host name of the VoIP Gateway (Optional)

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5. Click on Save Settings/Reboot on the left menu and then click the Save & Reboot button to save

your settings.

6. The NetComm ATA will reboot automatically to make your changes effective. Please wait while it reboots.

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DHCP Mode

1. Click on WAN on the left menu.

2. Change the IP Type to DHCP.

3. After that, click the Submit button.

Option Description

DNS Server 1

DNS Server 2

MAC

Host Name

4. Click on Save Settings/Reboot on the left menu and then click the Save & Reboot button to save

your settings.

The Ethernet MAC address of the WAN interface. (Optional)

The host name of the VoIP Gateway. (Optional)

5. The NetComm ATA will reboot automatically to make your changes effective. Please wait while it reboots.

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PPPoE Client Mode

1. Click on WAN on the left menu.

2. Change the IP Type to DHCP.

3. Enter the username and password supplied by your Internet Service Provider.

4. After that, click the Submit button.

Option Description

User Name

Password

Service Name

5. Click on Save Settings/Reboot on the left menu and then click the Save & Reboot button to save

your settings.

Your PPPoE username. This is normally speciﬁed by your Internet Service Provider (ISP).

Password for the above PPPoE account.

The name of your service provider. (Optional)

6. The NetComm ATA will reboot automatically to make your changes effective. Please wait while it reboots.

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VoIP Conﬁguration

1. Click on VoIP on the left menu.

2. Click on SIP Service Provider.

3. Enter the information supplied by your VoIP service provider and click Submit.

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Field Description

Active

SIP Proxy Domain

Proxy Server

Outbound Proxy

Display Name

User Name

Auth. Name

Auth. Password

Status

SIP Expiry Time

4. Click on Save Settings/Reboot on the left menu and then click the Save & Reboot button to save

your settings.

If set to on, the NetComm ATA will attempt to register with your VoIP service provider using this account.

The SIP Proxy Domain is speciﬁed by your VSP.

The host name of IP address of the SIP server. The proxy server is responsible for forwarding requests received from the NetComm ATA to your VoIP service provider. The Proxy Server is speciﬁed by your VoIP service provider.

The host name of IP address of the Outbound Proxy server. The Outbound proxy server is responsible for forwarding requests received from the NetComm ATA to your VoIP service provider. The Proxy Server is speciﬁed by your VoIP service provider.

The name or number the called party sees when you call them.

Your VoIP number. This number is assigned by your VoIP service provider.

The authentication username assigned by your VoIP service provider. This can be different to your VoIP number.

The authentication password provided by your VoIP service provider.

Indicates if the NetComm ATA is registered to your VSP. If “Registered” is shown, you can make and receive calls via your VoIP service provider. If the NetComm ATA reports “not registered”, check the SIP Service Provider settings above and try again.

This setting speciﬁes a recommendation to the SIP server on the re-registration interval. Increasing this time will send less registration trafﬁc, but can cause other issues.

5. The NetComm ATA will reboot automatically to make your changes effective. Please wait while it reboots.

6. Once the device has rebooted, check if the SIP LED is illuminated to indicate successful registration with

your VoIP Service Provider. If not, please check your settings.

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ADVANCED CONFIGURATION

Your NetComm ATA has many advanced features that you may want or need to use in the future. Let’s start by taking a look at the menus in the web interface.

1. Open your Web browser (Internet Explorer, Firefox etc).

2. Enter the LAN port default IP address of the NetComm ATA (e.g. http://192.168.22.1) in the address bar

and press Enter.

3. At the login screen, enter the gateway username and password. The default login username is admin,

and the default login password is admin.

4. On the Home Page, click on VoIP on the left menu to view the VoIP Gateway Conﬁguration page.

Many conﬁguration changes made using the web interface will only become active after clicking the Save & Reboot button on the Save Savings / Reboot page.

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Status Page

System Information Page

This page illustrate the system related information

Network Status Page

You can view the current Network setting in this page.

VoIP Status Page

The page shows current status of VoIP SIP Service provider.

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Conﬁguration

WAN

You can conﬁgure the WAN settings in this page.

The LAN mode allows the VoIP Gateway to be conﬁgured as a router or a bridge.

• Bridge: The system serves as a bridge between WAN port and LAN port.

• NAT: The system serves as a NAT router, allowing multiple computers and the VoIP gateway to share a

single public IP address.

The IP Type determines how the VoIP Gateway obtains its public IP address. Fixed IP allows you to manually enter a ﬁxed IP address, mask and gateway address. DHCP client allows the VoIP Gateway to discover an IP address automatically. You may need to refer to your current network environment to conﬁgure the VoIP Gateway properly.

The PPPoE setting is to setup the PPPoE Username and Password. If you have the PPPoE account from your Service Provider, please enter the Username and the Password correctly.

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LAN

You can conﬁgure the LAN settings/DHCP server in this page.

Options Description

Mask

MAC

DHCP Server

Start IP

End IP

Lease Time

The unique address of the VoIP Gateway on your local network.

The subnet mask of your local network.

The Ethernet MAC address of the WAN interface.

Enable or disable DHCP server.

The start of the IP address given by the DHCP server when the DHCP server is enabled.

The end of the IP address given by the DHCP server when the DHCP server is enabled.

The lease time for the DHCP address.

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VoIP

The VoIP Gateway Conﬁguration page sets parameters for all VoIP application.

The VoIP Gateway Conﬁguration page is divided into four general categories: SIP Setting, Phone Book, Phone Setting, and Others.

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SIP Setting

In the SIP settings sub menu you can setup the SIP Service Provider, Port Settings, Codec Settings, Codec ID Settings, DTMF Settings, RPort Setting and QoS.

SIP Service Provider

In the SIP Service Provider page you will need to enter the account and the related information for your SIP account. Please refer to your VSP (Voice Service Provider provider for more information. You can register two SIP accounts in the V220.

Options Description

Active

SIP Proxy Domain

Proxy Server

Outbound Proxy

Display Name

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If set to on, the VoIP gateway will attempt to register with your VoIP service provider using this account.

The SIP Proxy Domain is speciﬁed by your VSP.

The host name of IP address of the SIP server. The proxy server is responsible for forwarding requests received from the VoIP Gateway to your VoIP service provider. The Proxy Server is speciﬁed by your VoIP service provider.

The host name of IP address of the Outbound Proxy server. The Outbound proxy server is responsible for forwarding requests received from the VoIP Gateway to your VoIP service provider. The Proxy Server is speciﬁed by your VoIP service provider.

The name or number the called party sees when you call them.

User Name

Auth Name

Auth Password

Subscribe for MWI

Status

SIP Expiry Time

Use DNS SRV

Port Setting

You can conﬁgure the SIP and RTP port number in this page. Please leave as default unless otherwise speciﬁed by your VSP.

Your VoIP number. This number is assigned by your VoIP service provider.

The authentication username assigned by your VoIP service provider. This can be different to your VoIP number.

The authentication password provided by your VoIP service provider.

When enabled the Subscribe for Message Waiting Indication will be sent periodically.

Indicates if the VoIP Gateway is registered to your VSP. If “Registered” is shown, you can make and receive calls via your VoIP service provider. If the VoIP gateway reports “not registered”, check the SIP Service Provider settings above and try again.

This setting speciﬁes a recommendation to the SIP server on the re-registration interval. Increasing this time will send less registration trafﬁc, but can cause other issues.

When DNS SRV is switched on, the VoIP Gateway will look up DNS SRV records and register with the server with the highest priority. DNS SRV can provide redundancy if the primary SIP server becomes unreachable. Please check with your VSP to ﬁnd out if they support DNS SRV.

Options Description

SIP Port

RTP Port

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The port on the SIP server that receives requests from the ATA. Default: 5060.

The base port to receive RTP packets. Default: 60000.

Codec Settings

You can set the Codec Priority, RTP Packet Length, and VAD (Voice Activity Detection) settings in this page. Please leave as default unless otherwise speciﬁed by your VSP.

Codec ID Setting

You can set the value of Codec ID in this page. Please leave as default unless otherwise speciﬁed by your VoIP Service Provider.

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DTMF Setting

You can set the Out of Band DTMF and Send DTMF SIP Info Enable/Disable in this page. To change this setting, please follow your VoIP Service Provider’s information. When you ﬁnished the setting, please click the Submit button.

• RFC 2833: Click this button to send mid-call DTMF tones in RTP packets separately using RFC2833, i.e.,

dynamic negotiation of RTP payload for DTMF digits will be done.

• Inband DTMF (IN AUDIO): Click this button to send mid-call DTMF tones in RTP packets with the same

payload as voice, i.e., dynamic payload negotiation for DTMF digits will not be done.

• Send DTMF SIP Info: This ﬁeld is conﬁgurable when RFC 2833 is selected as the DTMF relay

mechanism. Specify the payload number that needs to be used for DTMF information negotiated in SDP during SIP signaling.

RPort Function

You can enable or disable the RPort feature in this page. To change this setting, please follow the recommendation of your VSP.

QoS

You can conﬁgure the Hold by RFC and Voice/SIP QoS in this page.

The QoS setting is to set the voice packets priority. If you set the value higher than 0, then the voice packets will get the higher priority to the Internet. But the QoS function still needs to cooperate with the others Internet devices.

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Phone Book Conﬁguration

The Phone Book contains Speed Dial Settings where you can setup Speed Dial number. If you want to use Speed Dial you just dial the speed dial number then press “#”.

The Phone Book has a maximum of 140 speed dial entries. You can add and delete the entry from this page.

To add a phone number into the Speed Dial list, enter the position, the name, the number (Speed Dial Number), and the phone number (by URL type) and then click the “Add Phone” button.

If you want to delete a phone number, you can select the phone number you want to delete then click “Delete Selected” button.

If you want to delete all phone numbers, you can click “Delete All” button.

Phone Book Page

Book Page

Phone

Name

Number

URL

Select

Delete Selected [Button]

Delete All [Button]

Reset [Button]

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Default page is Page1. There are total 14 pages from Page 1 to Page 14

Show the phone number by sequence. There are total 140 phone numbers from Phone 0 to Phone 139 can be set

Display the Name for the Speed Dial Number

Display the Speed Dial Number that you conﬁgured

Display the URL or number that you conﬁgured

Select the item.

Delete selected item

Delete all items

Reset selected item

Add New Phone

Position

Name

Number

URL

Add Phone [Button]

Reset [Button]

Examples:

Example 1: Position: 0, Name: “test iptel”, Number: 000, URL: test@iptel.org

When the user dials the Number 000, it will dial to the VoIP User test who registers to the SIP Server iptel.org.

Be noted that you need also register to the SIP Server iptel.org. If you register to the different SIP Server, please make sure that the SIP Server allows you dial to iptel.org.

Example 2: Position: 1, Name: “ip address 10.32”, Number: 001, URL: 192.168.10.32

Enter the phone number from 0 to 139

Enter the Speed Dial Name

Enter the Speed Dial Number

Enter the URL, VoIP Phone Number, Remote WAN IP Address of VoIP Gateway

Add the new Phone which you conﬁgured

Reset conﬁgured items

When the user dials the Number 001, it will dial to the VoIP Device whose IP Address is 192.168.10.32.

Example 3: Position: 2, Name: “ip address 10.32 5062”, Number: 002, URL: 192.168.10.132:5062

When the user dials the Number 002, it will dial to the VoIP Device whose IP Address is 192.168.10.132 with the port 5062.

Example 4: Position: 3, Name: “john”, Number: 003, URL: 88888888

When the user dials the Number 003, it will dial to the VoIP User whose phone number is 88888888.

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Example 5: Position: 4, Name: “Voip user”, Number: 004, URL: voipuser

When the user dials the Number 004, it will dial to the VoIP User whose phone number is voipuser.

Example 6: Position: 5, Name: “home”, Number: 005, URL: 0019998887777

When the user dials the Name [Speed Dial Number] 005, it will dial to the PSTN phone number 000019998887777 by VoIP OUT.

Note: please make sure that if your VoIP Service Provider support the VoIP OUT.

If your VoIP Service Provider supports the VoIP OUT, please follow the instructions of your VoIP Service Provider to dial the PSTN phone number by VoIP OUT.

For example: dialing as per the VoIP Server Provider recommended dialing sequence, 00 + country code + telephone number (e.g. 00 1 999 888 7777).

Example 7: When dial a VoIP Phone Number which isn’t conﬁgured in the Name [Speed Dial Number] list, it’ll dial out the VoIP Phone Number.

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Phone Setting

Phone Setting contains Call Forward, Volume Settings, DND Settings, Auto Answer, Caller ID, Dial Plan Settings, Flash Time Settings, Call Waiting Settings, T.38(FAX) Settings and Hot line Settings functions.

Call Forward function

You can setup the phone number you want to forward in this page. There are three type of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by enabling the respective mode.

Option Description

All Forward

Busy Forward

No Answer Forward

Off

IP/ON

PSTN (Optional)

All Fwd No.

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All incoming call will forward to the URL/number you conﬁgured.

If you are on the phone, the new incoming call will forward to the URL/number you conﬁgured.

If you can not answer the phone after a speciﬁc ring you conﬁgured, the incoming call will forward to the URL/number you conﬁgured.

Disable call forward.

Enable call forward for URL/number.

Enable call forward for PSTN phone number. Only for V211.

The URL/number you conﬁgured will be forward to for All Forward

Busy Fwd No.

No Answer Fwd No.

Name

URL

No Answer Fwd Time Out

Example 1: All Forward: IP, Name.: 7777, URL/Number: 7777

All incoming call will forward to the VoIP phone number 7777.

Example 2: All Forward: IP, Name: 192.168.10.36, URL/Number: 192.168.10.36

The URL/number you conﬁgured will be forward to for Busy Forward

The URL/number you conﬁgured will be forward to for No Answer Forward

Display the name of URL/number that you conﬁgured

Enter the URL, VoIP Phone Number, Remote WAN IP Address of VoIP Gateway which you want forward to.

You can set the Time Out time for system to start to forward the call to the number you conﬁgured for No Answer Forward

All incoming call will forward to the VoIP IP Gateway’s WAN IP Address 192.168.10.36.

Example 3: All Forward: PSTN, Name.: 88888888, URL/Number: 88888888

All incoming call will forward to the PSTN phone number 88888888.

Example 4: All Forward: IP, Name.: 7777, URL/Number: 7777

If you are on the phone, the new incoming call will forward to the VoIP phone number 7777.

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Example 5: All Forward: IP, Name: 192.168.10.36, URL/Number: 192.168.10.36

If you are on the phone, the new incoming call will forward to the VoIP IP Gateway’s WAN IP Address

192.168.10.36.

Example 6: All Forward: IP, Name.: 7777, URL/Number: 7777

If you can not answer the phone after 3 rings, the incoming call will forward to the VoIP phone number 7777.

Example 7: All Forward: IP, Name: 192.168.10.36, URL/Number: 192.168.10.36

If you can not answer the phone after 3 rings, the incoming call will forward to the VoIP IP Gateway’s WAN IP Address 192.168.10.36.

Example 8: All Forward: PSTN, Name.: 88888888, URL/Number: 88888888

If you can not answer the phone after 3 rings, the incoming call will forward to the PSTN phone number

88888888.

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Volume Setting function

You can setup the Handset Volume, PSTN-Out Volume, Handset Gain and the PSTN-In Gain.

Options Description

Handset Volume:

PSTN-Out Volume:

Handset Gain:

PSTN-In Gain:

DND Setting function

The do not disturb feature allows the device to automatically return busy during a speciﬁed time period. Incoming calls may return busy or go to voice mail dependant on the setup of your VSP account.

to set the volume for on your handset.

(V211 only) is to set the PSTN volume out.

to set the volume sent out to the other party handset.

(V211 Only) is to set the volume sent in to your handset from PSTN.

Options Description

DND Always

DND Period

From

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When enabled, incoming calls will be blocked and the caller will hear the busy tone.

When enabled, incoming calls will be blocked and the caller will hear the busy tone when the call is placed during the time period speciﬁed below. If the “From” time is larger than the “To” time, the block time will from day 1 to day 2.

Enter the start time of the time period. (24 hours format, hh:mm)

Enter the end time of the time period. (24 hours format, hh:mm)

Auto Answer function (Only applicable for V211)

You can set the Auto Answer function to make the device answer the incoming call.

If the call arrives from VoIP, then the VoIP Gateway can let the user redial the call to a PSTN phone number. If the call is coming from the PSTN, then the VoIP Gateway can let the user redial to a VoIP Phone number.

Once the VoIP Gateway receives the speciﬁed number of rings in the Auto Answer Counter and the caller has authenticated by entering the correct PIN code as conﬁgured, the caller can make a call on the other network. For example, you can call in on the VoIP network and make a PSTN call out on the FXO port or you can call in on the PSTN line (FXO port) and make a VoIP call.

In order to make a call, simply enter the desired number or speed dial number followed by the ‘#’ key.

If you call in via PSTN and make a VoIP toll bypass call, you can terminate the call by simply hanging up. If you call in via the VoIP network and make a PSTN toll bypass call, the call is terminated by ending the VoIP call.

Options Description

Auto Answer

Auto Answer Counter

PIN Code Enabled

PIN Code

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When enabled, the gateway will automatically answer the call.

Set the number of rings before the gateway answers the call.

Turn on to make the gateway prompt for a pin code before placing the call.

Enter the pin code required to be entered by the caller to make a call.

Example 1: Auto Answer: On, Auto Answer Counter: 3

How to Use – PSTN to VoIP Call:

1. Call in via PSTN.

2. When you hear the dial tone indicating that the VoIP Gateway is receiving 3 rings and expecting a number,

dial the VoIP phone number to which you want to call, then press # (optional) to make a PSTN to VoIP call.

How to Use – VoIP to PSTN Call:

1. Call in via VoIP.

2. When you hear the dial tone indicating that the VoIP Gateway is receiving 3 rings and expecting a number,

dial the VoIP phone number to which you want to call, then press # (optional) to make a VoIP to PSTN call.

Example 2: Auto Answer: On, Auto Answer Counter: 3, PIN Code Enabled: On, PIN Code: 1234

How to Use – PSTN to VoIP Call:

1. Call in via PSTN.

2. When you hear the continued BEEP BEEP indicating that the VoIP Gateway is asking you to enter the PIN Code:

3. Enter the correct PIN Code 1234, then press #.

4. When you hear the dial tone indicating that the VoIP Gateway is expecting a number, dial the VoIP phone

number to which you want to call, then press # (optional) to make a PSTN to VoIP call.

How to Use – VoIP to PSTN Call:

1. Call in via VoIP.

2. When you hear the continued BEEP BEEP indicating that the VoIP Gateway is asking you to enter the PIN Code:

3. Enter the correct PIN Code 1234, then press #.

4. When you hear the dial tone indicating that the VoIP Gateway is expecting a number, dial the PSTN phone

number to which you want to call, then press # (optional) to make a VoIP to PSTN call.

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Caller ID function

You can set the device to show Caller ID in your PSTN Phone or IP Phone. In Australia, Caller ID is sent using FSK after the 1st ring.

Options Description

Single Caller ID

CID Without Time

CID Type 2

Enable to detect Single Caller ID

Enable to detect the Caller ID without time.

Enable to detect the Caller ID Type 2.

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Dial Plan function

Dial plans can make your dialing easier. If the dialed number meets the preﬁx, the ATA will modify the dialed number before dialing it. If the dialed number does not meet the preﬁx, the ATA will dial the number without any modiﬁcation.

Options Description

Drop Preﬁx

Replace rule1 Replace rule2 Replace rule3 Replace rule4

Auto Dial Time

Symbol explain:

If set to yes, the ATA will remove the preﬁx from the number before passing it onto the VSP.

There are total 4 replace rules for use. +: or xxx: Deﬁne the length of digits.

Determines the time the ATA will wait before making the call after the last digit has been dialed.

x or X 0,1,2,3,4,5,6,7,8,9 (Wildcard)

+ Or

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Example 1: Drop preﬁx: No, Replace rule 1: 002, 8613+8662

When the number 8613 has been dialed, the preﬁx 002 will be added and the real phone number [002+8613+xxx] will be dialed out.

For example, when you dial the number 86315555 and the preﬁx 002 will be added and the real phone number 00286135555 will be dialed out.

When the number 8662 has been dialed, the preﬁx 002 will be added and the real phone number [002+8662+xxx] will be dialed out.

For example, when you dial the number 86625555 and the preﬁx 002 will be added and the real phone number 00286625555 will be dialed out.

Example 2: Drop preﬁx: Yes, Replace rule 2: 006, 002+003+004+005+007+009

When the number 002 has been dialed, the digits 002 will be replaced to 006 and the whole digits [006+xxx] will be dialed out.

For example, when you dial the number 0025555 and the digits 002 will be replaced to 006 and then the real phone number 0065555 will be dialed out.

When the number 003 has been dialed, the digits 003 will be replaced to 006 and the real phone number [006+xxx] will be dialed out.

For example, when you dial the number 0035555 and the digits 003 will be replaced to 006 and then the real phone number 0065555 will be dialed out.

When the number 004 has been dialed, the digits 004 will be replaced to 006 and the real phone number [006+xxx] will be dialed out.

For example, when you dial the number 0045555 and the digits 004 will be replaced to 006 and then the real phone number 0065555 will be dialed out.

When the number 005 has been dialed, the digits 005 will be replaced to 006 and the real phone number [006+xxx] will be dialed out.

For example, when you dial the number 0055555 and the digits 005 will be replaced to 006 and then real phone number digits 0065555 will be dialed out.

When the number 007 has been dialed, the digits 007 will be replaced to 006 and the real phone number [006+xxx] will be dialed out.

For example, when you dial the number 0075555 and the digits 007 will be replaced to 006 and then the real phone number 0065555 will be dialed out.

When the number 009 has been dialed, the digits 009 will be replaced to 006 and the real phone number [006+xxx] will be dialed out.

For example, when you dial the number 0095555 and the digits 009 will be replaced to 006 and then the real phone number 0065555 will be dialed out.

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Example 3: Drop preﬁx: No, Replace rule 3: 009, 12

When the number 12 has been dialed, the preﬁx 009 will be added and the whole digits [009+12+xxx] will be dialed out.

For example, when you dial the number 125555 and the preﬁx 009 will be added and the real phone number 009125555 will be dialed out.

Example 4: Drop preﬁx: No, Replace rule 4: 007, 5xxx+35xx+21xx

When the number 5xxx has been dialed, the preﬁx 007 will be added and the whole digits [007+5xxx] will be dialed out. Be note that the range of xxx is from 000 to 999.

For example, when you dial the number 5000 and the preﬁx 007 will be added and the real phone number 0075000 will be dialed out.

For example, when you dial the number 5999 and the preﬁx 007 will be added and the real phone number 0075999 will be dialed out.

When the number 35xx has been dialed, the preﬁx 007 will be added and the whole digits [007+35xx] will be dialed out. Be note that the range of xx is from 00 to 99.

For example, when you dial the number 3500 and the preﬁx 007 will be added and the real phone number 0073500 will be dialed out.

For example, when you dial the number 3599 and the preﬁx 007 will be added and the real phone number 0073599 will be dialed out.

When the number 21xx has been dialed, the preﬁx 007 will be added and the whole digits [007+21xx] will be dialed out. Be note that the range of xx is from 00 to 99.

For example, when you dial the number 2100 and the preﬁx 007 will be added and the real phone number 0072100 will be dialed out.

For example, when you dial the number 2199 and the preﬁx 007 will be added and the real phone number 0072199 will be dialed out.

When the number 534 have been dialed, the preﬁx 007 will not be added and the real phone number 534 will be dialed out due to 534 (3 digits) is not in the rule 5xxx (4 digits).

When the number 358822 have been dialed, the preﬁx 007 will not be added and the real phone number 358822 will be dialed out due to 358822 (6 digits) is not in the rule 35xx (4 digits).

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Auto Dial Time function

This value determines the time the ATA waits after the last digit is entered, before placing the call. Alternatively, you can press # to immediately dial the number.

Flash Time Settings function

When you use the PSTN phone and you need to press the Hook to simulate Flash key (Switch to the other phone line or HOLD), this function is to set the amount of time you need to press the Hook to represent the Flash function.

Call Waiting Setting

You can enable/disable the call waiting setting in this page.

When a new incoming call arrives while you are on an existing call, you can push the Flash button to switch to the new call. You can then use the Flash button to switch between the two calls.

To end the ﬁrst call, hang up the phone. The phone will now ring, pick it up to talk to the second caller. To end this call, simply hang up.

T.38 (FAX) Setting

You can enable/disable the T.38 FAX function in this page.

Options Description

T.38 (FAX)

T.38 Port

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Switch on to enable T.38 fax support.

Default is 60000.

T.38 support

Fax Pass-through

In fax pass-through mode, UDPTL packets are not used. Fax communication between the two fax machines is carried out entirety in-band over a voice call (over RTP). The ATA is aware that the call in progress is a fax call and not voice call. If during a voice call, the CED/CNG fax tones are recognized, then the VoIP Gateway will change the voice codec to G.711, and if necessary, turn off echo cancellation (EC) and voice activity detection (VAD) and adjust the jitter and reorder buffers to ﬁx the network delay for the duration of the call.

T.38 support mode

T.38 provides an ITU-T standards-based method and protocol for fax. In this mode, the ATA will establish a normal voice call and switch to T.38 fax based on the detection of fax tones. This will cause the ATA to renegotiate the session parameters with new T.38 parameters. The rest of the fax signaling and data is then encapsulated and sent in IFP packets. The ITU-T T.38 deﬁnes the behavior for both Internet Aware Fax Devices (IAF, network aware fax machine) and gateways connected to G3FE (Group 3 Fax equipment). This ATA supports both kinds of behaviors.

Hot line Settings

The hot line function causes the ATA to dial the hot line number speciﬁed immediately after the phone is picked up.

Options Description

Use Hot Line

Hot Line Number

Example 1: Use Hot Line: Enable, Hot line number: 2468013579

Every time when you pick up the phone, it will dial to the VoIP Phone Number 2468013579 automatically.

Example 2: Use Hot Line: Enable, Hot line number: 192.168.10.63

Every time you pick up the phone, it will establish a call to the SIP device at the IP address 192.168.10.63.

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Select enable to enable the Hot Line function.

Enter the URL, VoIP Phone Number or Remote WAN IP Address of VoIP device which you want to use for Hot Line.

Others

The others menu contains the following sub menus :

Auto Conﬁguration Settings

You can enable/disable the auto conﬁguration/provisioning setting in this page.

The ATA allows for provisioning and remote upgrade. Provisioning is achieved through conﬁguration proﬁles transferred to the device via TFTP, HTTP or FTP. The ATA can be conﬁgured to update its VoIP Conﬁguration from a remote proﬁle on power up or reboot.

Options Description

Auto Conﬁguration

TFTP Server

HTTP Server

HTTP Path

FTP Server

FTP Username

FTP Password

File Path

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Enable Auto Conﬁguration. (Default:off)

Enter IP or Host Name of TFTP Server.

Enter IP or Host Name of HTTP Server.

Enter the path where the provisioning ﬁle is located.

Enter IP or Host Name of FTP Server.

Enter Username which provided by FTP Server.

Enter Password which provided by FTP Server.

Enter the path where the provisioning ﬁle is located.

Example 1: Auto Conﬁguration for HTTP Server

Auto Conﬁguration: HTTP, HTTP Server: 192.168.10.100, HTTP Path: /

Every time when you power on the ATA it will update its VoIP conﬁguration to the latest one from Auto Provisioning Server (HTTP Server) automatically.

Example 2: Auto Conﬁguration for TFTP Server

Auto Conﬁguration: TFTP, TFTP Server: 192.168.10.100

Every time when you power on the ATA, it will update its VoIP conﬁguration to the latest one from Auto Provisioning Server (TFTP Server) automatically.

Example 3: Auto Conﬁguration for FTP Server

Auto Conﬁguration: FTP, FTP Server: 192.168.10.100, FTP Username: 1234, FTP Password: 1234, FTP Path: /

Every time when you power on the VoIP Gateway, it will update its VoIP conﬁguration to the latest one from Auto Provisioning Server (FTP Server) automatically.

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FXO & FXS Impedence Setting

You can select the FXO & FXS Impedence Setting for a different country. Please leave as default unless otherwise instructed.

Options Description

FXO Port

FXS Port

Select Impedence for the FXO port. (only on V211)

Select Impedence for the FXS port.

STUN Setting

You can enable/disable the STUN feature and conﬁgure the STUN Server IP address in this page. This function can help your ATA to work properly behind a NAT device. To change these settings please follow your VoIP Service Provider’s information.

Options Description

STUN

When enabled the NetComm ATA uses STUN (Simple Transversal of UDP through NAT) if the unit is behind a NAT enabled router and the router has no ALG for SIP, or NONE to disable STUN (VoIP Gateway is not to use STUN for NAT traversal). VoIP Gateway also supports a proprietary implementation of NAT traversal where the Service provider is expected to provide some relay support. If NONE is selected, then based on the responses received, the VoIP Gateway will dynamically determine if the SIP Server supports the proprietary implementation.

Note: Even when STUN is enabled, the VoIP Gateway does an automatic detection of the presence of SIP ALG and disables the use of STUN. This is to avoid some media problems arising out of the behavior of some ALGs when STUN is used at the user end.

STUN Server

STUN Port

Enter the IP address or host name of the STUN server.

Enter the port number on which the STUN server listens for requests from the ATA. (default : 3478)

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MAC Clone Settings

Some ISPs have a restriction that allows only 1 host locked by its MAC address to talk on the internet. If you change network cards, you have to call them up to change the MAC. The NetComm ATA can clone the computer’s MAC that was originally set up in this conﬁguration.

Options Description

MAC Clone

Advanced Settings

Default is Off (disable). When it was On (enable). The VoIP Gateway clones the computer’s MAC that was originally set up for such an ISP.

Options Description

ICMP Not Echo

Send Anonymous CID

Management from WAN

Billing Signal

CPC Delay

CPC Duration

Send Flash event

SIP Encrypt

PPPoE retry period

System Log Server

System Log Type

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When switched to yes, the ATA will not echo the ICMP request.

Send Anonymous as the Caller ID to display when you make a VoIP call.

Switch to yes to enable remote management.

To enable billing signal. Leave as default unless instructed by your VoIP Service Provider.

Default is 2. The VoIP Gateway will send the CPC after the delay time which you conﬁgure.

When VoIP Gateway is the called party, CPC duration is the “voltage drop” duration, before it plays dial tone again.

Default is Disable. There are two types of Flash event: DTMF Event and SIP Info.

Default is Disable. There are four types of SIP Encrypt: INFINET, AVS, WALKERSUN1 and WALKERSUN2.

Time before the PPPoE client retries the connection.

IP address of the System log server

Type of system log generated.

DDNS

Dynamic DNS (DDNS) allows the ATA to register your IP address with a DDNS provider each time the IP address changes. This allows you to discover the ATA from a remote location when the ATA is used with a dynamically assign IP address service.

Example:

1. Conﬁgure the WAN connection to PPPoE Client and make sure the ATA has the WAN IP Address (Public IP

Address).

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2. Enable DDNS and conﬁgure the DDNS settings including Host Name, User Name and Password

3. If conﬁgured correctly, you should be able to visit the home page of the ATA by entering the DDNS Host

Name as follow.

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VLAN Settings

You can set the VLAN settings in this page.

Options Description

VLAN Packets

VID (802.1Q/TAG)

User Priority (802.1P)

CFI

When enabled the ATA will receive VLAN Packets function.

Conﬁgure the Virtual LAN ID (VLAN ID or VID) for VLAN Server.

The VLAN Identiﬁer is a 12-bit ﬁeld. It uniquely identiﬁes the VLAN to which the frame belongs. The ﬁeld can have a value between 2 and 4094. (Default 136).

Default is 0. Conﬁgure user priority.

Also known as user priority, this 3-bit ﬁeld refers to the IEEE 802.1p priority. The ﬁeld indicates the frame priority level which can be used for the prioritization of trafﬁc. The ﬁeld can represent 8 levels (0 through 7).

The Canonical Format Indicator is a 1-bit ﬁeld.

If the value of this ﬁeld is 1, the MAC address is in non-canonical format. If the value is 0, the MAC address is in canonical format.

DMZ

A DMZ Host PC is set up ‘between’ your (private) LAN and the (public) WAN to allow access from the outside world to a speciﬁed and isolated zone on your network. It is most commonly used to provide access to a Web server or Game server without exposing the rest of your computers to the Internet. Turn on the DMZ and enter the IP address of the DMZ computer and click Submit. The computer with that IP address can then serve web pages or games to the outside world, while the rest of your network remains private.

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Virtual Server

Virtual Servers are used for port forwarding from the WAN to LAN networks. If any local PCs need to be mapped to the UDP/TCP port on WAN side, you can enter the mappings here. There can be up to 24 different Virtual Server conﬁgurations.

Options Description

Virtual Server Page

Num

Enable

Protocol

In Port (Internal Port)

Ex Port (External Port)

Server IP

Select

Enable Selected [Button]

Delete Selected [Button]

Delete All [Button]

Reset [Button]

Default page is Page1. There are 3 pages, Page 1 to Page 3

Show the number by sequence. There are a total of 24 numbers from mapping 0 to mapping 23 that can be set.

Check to enable the port forwarding protocol.

TCP or UDP.

Internal port number on the LAN side.

External port number on the WAN side.

Display the private network IP address for the particular server.

Select the item of the Virtual Server

Enable selected item

Delete selected item

Delete all items

Reset selected item

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Add Virtual Server

Num

Server IP

Protocol

Internal Port

External Port

Add Server [Button]

Reset [Button]

Example 1 (FTP Server):

Num: 0, Server IP: 10.0.0.150, Protocol: TCP, Internal Port: 21, External Port: 21

Other people can visit your FTP Server by entering the WAN IP Address of the ATA and then the NetComm ATA will re-directly it to your LAN IP 10.0.0.150.

Enter the number corresponding to the Virtual Server conﬁguration.

Enter the private network IP address for the particular server.

Select TCP or UDP.

Enter the port number of the Private Network (LAN or internal network). In most cases, the private port number is same as public port number. This port number cannot be seen from the WAN side.

Enter the port number of the Public Network (WAN or external network).

Add the new Server which you conﬁgured

Reset conﬁgured items

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Well Known TCP/UDP Ports

Port Protocol UDP TCP

20 File Transfer Protocol (FTP) Data X

21 FTP Commands X

23 Telnet X

25 SMTP X

43 Whois X

53 Domain Name System (DNS) X X

69 Trivial File Transfer Protocol (TFTP) X

70 Gopher X

79 Finger X

80 HTTP X

110 POP3 X

111 SUN Remote Procedure Call (RPC) X

115 SFTP X

119 Network News Transfer Protocol (NNTP) X

123 Network Time Protocol (NTP) X

144 News X X

161 Simple Network Management Protocol (SNMP) X

162 SNMP traps X

179 Border Gateway Protocol (BGP) X

443 Secure HTTP (HTTPS) X

513 rlogin X

514 rexec X

517 talk X X

518 ntalk X X

520 Routing Information Protocol (RIP) X

1701 Layer 2 Tunneling Protocol (L2TP) X

2000 Open Windows X X

2049 Network File System (NFS) X

6000 X11 X X

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SNTP Settings

Simple Network Time Protocol (SNTP) can be used to sync the time on the NetComm ATA. If SNTP is enabled, your ATA clock is synchronized with an Internet time server once a day. However, if you don’t have a continuous Internet connection through a cable modem or DSL modem, the automatic synchronization might not always occur.

Options Description

Primary Server

Secondary Server

Time Zone

Sync

If time synchronization fails, it might be for one of the following reasons:

• You are not connected to the Internet. Establish an Internet connection before you attempt to synchronize

your clock.

• Your personal or network ﬁrewall prevents clock synchronization. Most corporate and organizational

ﬁrewalls will block time synchronization

• The time shown on your ATA is too different from the current time on the Internet time server. Internet

time servers might not synchronize your clock if your ATA’s time is off by more than 15 hours.

Enter the primary NTP server name or IP address.

Enter the secondary NTP server name or IP address. The secondary server will be used if the primary server becomes unreachable.

Enter the offset from GMT for your time zone.

Reports the synchronized time.

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Alarm Settings

Provide the alarm function. The alarm will sound when it reached the Alarm Time that you conﬁgured.

Options Description

Alarm

Alarm Time

Current time

Example 1: Alarm: ON, Alarm Time: 8:1(hh:mm)

Default is OFF (Disable). When it was ON (Enable), It will enable the Alarm function.

Default is 0:0 (hh:mm). Set the Alarm Time. (24 hours format, hh:mm)

The current time of the NetComm ATA.

The alarm will sound when it reached the current time 08:01.

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System Authority

In the System Authority page you can change the NetComm ATA’s login name and password.

Save Settings/Reboot

If modiﬁcations are made to the Netcomm ATA’s settings, they will be lost if they are not saved to the ATA nonvolatile memory. Click Save & Reboot button to make any conﬁguration changes permanent. After you click the Save & Reboot button, the ATA will automatically restart and the new setting will take effect.

If you only want to reboot the NetComm ATA, Click the Reboot Only button. After you click the Reboot Only button, the ATA will automatically restart.

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System

Reset factory default Page

You can restore the NetComm ATA settings to factory default by clicking the restore button. The ATA will restore to default and automatically restart.

Firmware Update Page

To update the NetComm ATA ﬁrmware, click browse to locate the new ﬁrmware or enter the location and name of the ﬁrmware in the File Location ﬁeld. Click Update, to upgrade the ﬁrmware to the new version.

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Auto Update Page

The NetComm ATA can check and update the ﬁrmware automatically either when powering up or via a schedule.

Options Description

Update via

TFTP Server

HTTP Server

HTTP File Path

FTP Server

FTP Username

FTP Password

FTP File Path

Check new ﬁrmware

Scheduling (Date)

Scheduling (Time)

Enable or disables the Auto Update function. To enable, select the type of auto update server – TFTP, FTP or HTTP.

The address of the TFTP.

The address of the HTTP server.

The path where the ﬁle is located.

The address of the FTP server.

The username for the FTP server.

The password for the FTP server.

The path where the ﬁle is located.

Determines when the NetComm ATA will check for new ﬁrmware. When power ON is selected the ATA will check for updates when powering up. If scheduling is selected, the VoIP gateway will check for updates using the schedule set below.

The VoIP Gateway will check if there is new ﬁrmware on the TFTP/FTP/HTTP Server periodically. The range of the Scheduling Date is 1 - 30. Default is 14.

Default is AM 00:00- 05:59. The VoIP Gateway will check if there is new ﬁrmware on the TFTP/FTP/HTTP Server periodically. There are four Scheduling Time: AM 00:0005:59, AM 06:00- 11:59, PM 12:00- 17:59, PM 18:00- 23:59

V110, V210P, V211, V220 User Guide YML881Rev1 58 www.netcomm.com.au

Automatic Update

Firmware File Preﬁx

Next update time

Notify only: When newer ﬁrmware is available, the VoIP Gateway will only notify you by a “BEEP BEEP BEEP” you when you pick up the phone. It will not automatically upgrade.

Automatic (Scheduling): When there is a newer ﬁrmware, it will update the ﬁrmware automatically.

The ﬁle preﬁx of the ﬁrmware

Displays the next update or check time.

Notify Only Auto Update Procedure:

1. Power on the VoIP Gateway and it will check if there is any newer ﬁrmware available on the Server. When

there is a newer ﬁrmware, it will only notify you by “BEEP BEEP BEEP” after you pick up the phone.

2. Please press #190# and then hang up the phone to unlock the special key on keypad.

3. Please press #160# and then hang up the phone to ask NetComm ATA to update the ﬁrmware

immediately.

4. It takes around 7 minutes for updating the new ﬁrmware.

AUTO UPDATE PROCEDURES:

1. Power on the NetComm ATA and reaches Next update time. It will check if there is any update ﬁrmware

is newer one on the Server. When there is a newer ﬁrmware, it’ll update the ﬁrmware automatically. It will take around 7 minutes to update the ﬁrmware.

Be noted:

1. If the NetComm ATA is powered off and has passed the Next update time, it will not update the ﬁrmware

after you power on the NetComm ATA. It’ll only update when the NetComm ATA is power on and reaches Next update time.

2. If you are on the phone and have a conversation to others by VoIP and the Next update time is passing, it

will update the ﬁrmware immediately after you hang up the phone.

YML881Rev1 V110, V210P,V211, V220V110, V210P, V211, V220 User Guide www.netcomm.com.au 59

IVR Interface for VoIP Gateway

You can use the PSTN phone to conﬁgure the NetComm ATA. Please follow the instruction below to conﬁgure it.

Group IVR Action IVR Menu Choice Parameter(s) Notes:

Function Reboot #195# None After you hear “Option

Successful,” hang-up. The system will reboot automatically.

Function Factory Reset #198# None System will automatically

Reboot. WARNING: ALL “UserChangeable” NON DEFAULT SETTINGS WILL BE LOST! This will include network and service provider data.

Info Check IP Address #120# None IVR will report the LAN port IP

address

Info Check IP Type #121# None IVR will report the WAN Port

DHCP is enabled or disabled.

Info Check the Phone

Number

Info Check Network

Mask

Info Check Primary DNS

Server Setting

Info Check IP Address #126# None IVR will report the WAN port IP

Info Check Firmware

Version

Setting Set DHCP client #111# None The system will change to DHCP

Setting Set Static IP Address #112xxx*xxx*xxx*xxx# Enter IP address using

Setting Set Network Mask #113xxx*xxx*xxx*xxx# Must set Static IP ﬁrst.

Setting Set Gateway IP #114xxx*xxx*xxx*xxx# Must set Static IP ﬁrst.

Setting Set Primary DNS

Server

Setting Set Codec #130+[1-8]# 1:G.711 u-Law, 2: G.711 a-

Set Handset Gain #131+[00~15]# Handset Gain from 0~15 You can set the Handset gain to

Set Handset Volume #132+[00~12]# Handset Volume from 0~12 You can set the Handset volume

#122# None IVR will report the VoIP number

in use.

#123# None IVR will report the WAN Port

network mask

#125# None IVR will announce the current

setting in the Primary DNS ﬁeld.

address

#128# None IVR will announce the version of

the ﬁrmware running on the VoIP Gateway.

client type

DHCP will be disabled and numbers on the telephone key pad. Use the *(star) key when entering a decimal point.

#115xxx*xxx*xxx*xxx# Must set Static IP ﬁrst.

Law, 3:G.723.1, 4: G.729a, 5: G.726 16K, 6:G.726 24K, 7: G.726 32K, 8: G.726 40K,

system will change to the Static

IP type.

You can set the codec you want

to the ﬁrst priority.

proper value, default is 6

to proper value, default is 10

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How to make a phone call

Dial a VoIP Phone call

When your VoIP Gateway is conﬁgured properly, you can make a phone call via VoIP.

If you want to make a phone call, you can dial the phone number and press “#” button to start to dial the phone number.

Dial a PSTN Phone call (V210P/V211)

By default outgoing calls should be placed with the VoIP service. If you want to make a phone call via the PSTN pass-through port, you can press “0*” and dial the phone number.

Note: On loss of power, the V210P/V211 will switch all calls from the FXS port through to the PSTN pass-through.

The VoIP Gateway also provides some functions that list as below:

Call Features

The NetComm ATA also provides some functions that list as below:

Blind Transfer

This feature allows a user (transferor) to transfer an existing call to another telephone number (transfer target) without connecting to the transfer target number.

How to Use:

1. During an existing call, perform a hook ﬂash to put the other party on hold and get a dial tone.

2. When you hear the dial tone, press #510# on your telephone dial-pad.

3. When you hear the dial tone indicating that the VoIP Gateway is expecting a number, dial the phone

number to which you want to transfer the other party, then press # (optional) and then hang up the phone.

Attendant Transfer

This feature allows a user to transfer an existing call to another telephone number after ﬁrst consulting with the dialed party (transfer target) before hanging up.

How to Use:

1. During an existing call, perform a hook ﬂash to put the other party on hold and get a dial tone.

2. When you hear the dial tone, press #511# on your telephone dial-pad.

3. When you hear the dial tone, dial the telephone number to which the existing party is to be transferred,

then press # (optional).

4. When the target transfer answers the phone, you may consult with the target transfer, and then hang up

your phone to transfer the call to the target transfer.

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3-Way Conferencing

How to Use:

1. Dial the ﬁrst number.

2. During connection to the ﬁrst party, perform a hook ﬂash to put the ﬁrst party on hold.

2. When you hear the dial tone, press #512# on your telephone dial-pad.

3. When you hear the recall dial tone, dial another number and talk with the second person.

4. To conference with both callers at the same time, perform a hook ﬂash.

5. To transfer the second call to ﬁrst call, perform a hook ﬂash after entering into conferencing mode.

Note: If you hang up during conferencing, it will transfer the ﬁrst call to the second call.

Call Waiting

How to Use:

1. When a new call is coming while you are talking, you can push the Flash button or perform a hook ﬂash

to switch to the new call.

2. You can push the Flash button to switch between the two calls.

1. Dial the ﬁrst number to make a conversation.

2. During connection to the ﬁrst party, push the Flash button or perform a hook ﬂash to put the ﬁrst party on

hold.

3. When you hear the dial tone, dial another number and talk with the second person.

4. You can push the Flash button or perform a hook ﬂash to switch between the two calls.

Call Hold

How to Use:

1. When a new call is coming in while you are on an existing call, you can push the Flash button or perform

a hook ﬂash to hold the current call for a while, then push the Hold key again to keep talking.

2. You can push the Flash button to switch between the two calls.

V110, V210P, V211, V220 User Guide YML881Rev1 62 www.netcomm.com.au

COMMON PROBLEM AND SOLUTIONS

No power to the ATA.

• Please ensure that the power cable is connected correctly and

• The power switch is turn on. (Both at the power point and the power switch at the back of the unit).

Note: Only use the power adapter supplied with the NetComm ATA. A different adapter may damage the product.

ETH light is off.

• Please ensure that the ATA is connected to your broadband modem/router using the correct cable and

your computer is connected to the ATA.

No Dial Tone and the SIP light is off. Please check the following:

• The WAN port of the unit is conﬁgured correctly according to your network structure and

• The SIP is conﬁgured correctly according to your VSP (VoIP Service Provider). You have to ensure that all

details are entered correctly.

There is no sound coming from the phone handset.

• Please ensure that you phone handset is connected to the correct port (phone port) and the cable is not

damage.

When making a call, the call does not go through and always come with busy

tone.

• Check that the SIP light is on.

• Try to call a different number.

• If your broadband mode/router is not VoIP aware, you must set up port forwarding to allow the voice

trafﬁc to go through the modem/router. There are certain ports that you need to specify, 5060-5061 (UDP) and 60000-60008 (UDP). Please consult your modem/router user guide for more information on how to conﬁgure port forwarding.

The ATA never receives call and all incoming calls go directly to voice mail.

• Check that the SIP light is on.

• Try to call from a different number.

• If your broadband mode/router is not VoIP aware, you must set up port forwarding to allow the voice

• Ensure that your VoIP account supports incoming call from a PSTN number. Depending on your VSP, not

all VoIP account supports incoming calls from a PSTN/mobile number.

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APPENDIX A: GLOSSARY

This glossary deﬁnes acronyms and keywords used in this document.

ATA Analog Telephone Adaptor

Codec The format by which audio or video streams are compressed for transmission over networks.

CPC CPC (Calling Party Control) is a signal sent from most modern electronic COs to indicate that the Calling Party

DTMF Dual-tone multifrequency. DTMF is the system that is used in interactive voice-response menu systems such as

FoIP Fax over Internet Protocol

FXO Foreign Exchange Ofﬁce

FXS Foreign Exchange Subscriber

IP Internet Protocol. A data-oriented protocol used for communicating data across a network. IP is the most

IP address A unique number that devices use in order to identify and communicate with each other on a computer network

MWI Message Waiting Indicator. An indicator that there is a voicemail message for the owner of an account.

PSTN Public Switch Telephone Network. The traditional land-line phone network.

RTP Real-time Transport Protocol. A protocol for delivering the media portion of a data transmission over an IP

Signaling In a VoIP phone call, the information in a call that deals with establishing and controlling the connection, and

SIP Session Initiation Protocol. The signaling protocol followed by VoIP Gateway for handling phone calls.

SIP account An account that provides the user the ability to make VoIP phone calls. The account encapsulates the rules and

SIP address The address used to connect to a SIP endpoint. In other words, the “phone number” used in a VoIP phone call.

STUN Simple Transversal of UDP through NAT

TCP Transmission Control Protocol. A transport protocol for delivering data over an IP network. Other transport

TLS Transport Layer Security. A transport protocol for delivering data over an IP network. TLS is a secure transport

UA User Agent

UDP User Datagram Protocol. A transport protocol for delivering data over an IP network. Other transport protocols are

URI URI Uniform Resource Identiﬁer. A name or address that identiﬁes a location on the world wide web. A SIP

URL Uniform Resource Locator. A URI that both identiﬁes a name or address and indicates how to locate it.

VoIP Voice over Internet Protocol. A variation of IP used for sending voice data over the internet, in other words, used

VoIP Service Provider A business that provides a VoIP service, allowing a user to connect to the internet in order to make VoIP phone

has hung up.

the menu system for accessing voicemail messages. The DTMF system allows the user to interact with the menu by pressing keys on the key pad.

common protocol used on the internet.

using the IP standard.

network. SRTP is another media protocol.

managing the network. The non-signaling portion of the call is the Media.

functions the user can access.

For example, sip:test@domainA.com.

protocols are TLS and UDP.

protocol, which means that all the data being transmitted (signaling and media) is encrypted. Other transport protocols are TCP and UDP.

TCP and TLS.

address is a type of URI.

for making phone calls over the internet.

calls using VoIP Gateway. The VoIP service provider sets up a SIP account for the user.

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PRODUCT WARRANTY

The warranty is granted on the following conditions:

1. This warranty extends to the original purchaser (you) and is not transferable;

2. This warranty shall not apply to software programs, batteries, power supplies, cables or other acces¬sories

supplied in or with the product;

3. The customer complies with all of the terms of any relevant agreement with NetComm and any other

reasonable requirements of NetComm including producing such evidence of purchase as NetComm may require;

4. The cost of transporting product to and from NetComm’s nominated premises is your responsibility; and,

5. NetComm does not have any liability or responsibility under this warranty where any cost, loss, injury or

damage of any kind, whether direct, indirect, consequential, incidental or otherwise arises out of events beyond NetComm’s reasonable control. This includes but is not limited to: acts of God, war, riot, embargoes, acts of civil or military authorities, ﬁre, ﬂoods, electricity outages, lightning, powersurges, or shortages of materials or labour.

6. The customer is responsible for the security of their computer and network at all times. Security features

may be disabled within the factory default settings. NetComm recommends that you enable these features to enhance your security.

The warranty is automatically voided if:

1. You, or someone else, use the product, or attempts to use it, other than as speciﬁed by NetComm;

2. The fault or defect in your product is the result of a voltage surge subjected to the product either by the way

of power supply or communication line, whether caused by thunderstorm activity or any other cause(s);

3. The fault is the result of accidental damage or damage in transit, including but not limited to liquid spillage;

4. Your product has been used for any purposes other than that for which it is sold, or in any way other than in

strict accordance with the user manual supplied;

5. Your product has been repaired or modiﬁed or attempted to be repaired or modiﬁed, other than by a qualiﬁed

person at a service centre authorized by NetComm; and,

6. The serial number has been defaced or altered in any way or if the serial number plate has been removed.

Limitations of Warranty

The Trade Practices Act 1974 and corresponding State and Territory Fair Trading Acts or legalisation of another Government (“the relevant acts”) in certain circumstances imply mandatory conditions and warranties which cannot be excluded. This warranty is in addition to and not in replacement for such conditions and warranties.

To the extent permitted by the Relevant Acts, in relation to your product and any other materials provided with the product (“the Goods”) the liability of NetComm under the Relevant Acts is limited at the option of NetComm to:

• Replacement of the Goods; or

• Repair of the Goods; or

• Payment of the cost of replacing the Goods; or

• Payment of the cost of having the Goods repaired.

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NOTES

________________________________________

V110, V210P, V211, V220 User Guide YML881Rev1 66 www.netcomm.com.au

Did you know your phones will disrupt

your ADSL2+ connection...

Even If You Have Older ADSL Filters?

Due to faster ADSL speeds, an inline ADSL2+ Microﬁlter stops your ADSL connection being disrupted by telephones connected to the same line. Older ﬁlters cannot handle these speeds. A high-quality Microﬁlter from NetComm will ensure you have a stable broadband Internet connection with no reduction in the quality of your telephone service.

Having one easy to install Microﬁlter for every telephone type device, including dial up modems and Foxtel, is essential to reducing voice and data service problems and interruptions.

NetComm EM1550 ADSL2+ Microﬁlters conform to relevant ACA requirements and Telstra speciﬁcations and will:

• Help you obtain faster technical support from your ISP

• Minimise unexplained disruption of your ADSL connection

• Work with all current ADSL connection speeds

• Microﬁlters are also splitters which allow you to connect the line between phone and modem

Grab your NetComm Microﬁlters today, either:

Visit a retailer closest to you...

Harvey Norman, Ofﬁceworks or Harris Technology.

YML881Rev1 V110, V210P,V211, V220V110, V210P, V211, V220 User Guide www.netcomm.com.au 67

Order direct...

Visit www.netcomm.com.au/ADSL/EM1550.php

Phone (02) 9424-2055 (Quote Ref No - 1550)

Product Warranty

NetComm products have a standard 12 months warranty from date of purchase. However some products have an extended warranty option, via registering your product online at the NetComm website www.netcomm.com.au. Refer to the User Guide for complete product warranty conditions, limitations of warranty and other legal and regulatory information.

Contact Information

If you have any technical difﬁculties with your product, please do not hesitate to contact NetComm’s Customer Support Department.

Email: support@netcomm.com.au

www.netcomm.com.au

Note: NetComm Technical Support for this product only covers the basic installation and features outlined in the Quick Start Guide. For further information regarding the advanced features of this product, please refer to the conﬁguring sections in the User Guide or contact a Network Specialist.

NetComm Limited ABN 85 002 490 486 PO Box 1200, Lane Cove NSW 2066 Australia E – sales@netcomm.com.au W – www.netcomm.com.au

V110, V210P, V211, V220 User Guide YML881Rev1 68 www.netcomm.com.au

NetComm V110 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual