Digicom VoiceGate ATA User Manual

VoiceGate ATA

User’s Guide

rev. 1.0 10/2007

VoIP Gateway Ethernet

2 VoIP lines

VoIP Gateway Configuration

1.0 Features

Network Protocol

Tone

SIP v1 (RFC2543), v2(RFC3261)

IP/TCP/UDP/RTP/RTCP

IP/ICMP/ARP/RARP/SNTP

TFTP Client/DHCP

Client/ PPPoE Client

Telnet/HTTP Server

DNS Client

NAT/DHCP Server

Ring Tong

Ring Back Tone

Dial Tone

Busy Tone

Programming Tone

Codec

Phone Function

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

G.723.1: 6.3k / 5.3k bit/s

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

G.729A: 8k bit/s (CS-ACELP)

G.729B: adds VAD & CNG to G.729

Volume Adjustment

Speed dial key

Phone book

Flash

Voice Quality

IP Assignment

VAD: Voice activity detection

CNG: Comfortable noise generator

LEC: Line echo canceller

Packet Loss Compensation

Adaptive Jitter Buffer

Static IP

DHCP

PPPoE

Call Function

Security

Call Hold

Call Waiting

Call Forward

Caller ID

3-way conference

HTTP 1.1 basic/digest authentication for Web setup

MD5 for SIP authentication (RFC2069/ RFC 2617)

DTMF Function

NAT Traversal

In-Band DTMF

Out-of Band DTMF

SIP Info

STUN

SIP Server

Configuration

Registrar Server, Outbound Proxy

Web Browser , Console/Telnet,IVR/Keypad

Firmware Upgrade

Auto Provisioning

TFTP, Console, HTTP

HTTP, FTP, TFTP

Interface

Modem & Fax modes

1 WAN port interface

1 LAN port interface

2 VOIP port interface (FXS)

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

detection

T.38 support

P

ARA METERS TH AT Y OU NEED TO CONFIG URE THE VOIP GAT EWAY

Following table is the parameters that you need to configure the VoIP Gateway.

If you cannot get the Internet/WAN access of your own network and VoIP Configuration of your VoIP Service

Provider, it’s difficult to configure the VoIP Gateway correctly and have it work properly.

Parameters that you need to configure the VoIP Gateway

Internet/WAN Access of your own Network

DHCP Client

PPPoE Client

Static IP

Obtain an IP Address automatically

X

N/A

N/A

Username

N/A

1234

N/A

Password

N/A

1234

N/A

IP Address

N/A

N/A

192.168.10.110

Subnet Mask

N/A

N/A

255.255.255.0

Gateway

N/A

N/A

192.168.10.100

DNS Server IP

N/A

N/A

192.168.10.100

VoIP Configuration of Your VoIP Service Provider

Domain Server Address

192.168.10.100

Domain Server Port

5060

Proxy Server Address

192.168.10.100

Proxy Server Port

5060

Outbound Proxy Address

192.168.10.100

Outbound Proxy Port

5060

Note:
* Username / Password which was given by Telecom or by your Internet Service Provider
(ISP).

* IP Address / Subnet Mask / Gateway / DNS Server IP which was given by your network
administrator or by Telecom or by your Internet Service Provider (ISP).

* Domain Server Address and Port / Proxy Server Address and Port / Outbound Proxy
Address and Port / User Name / Register Name / Register Password which was given by
Telecom or by your Internet Service Provider (ISP) or by your VoIP Service Provider.

2. VoIP Gateway Overview

VoIP Gateway has many ports, switches and LEDs. VoIP Gateway may have some or all of the features listed

below

2.1 Ports and Buttons

POWER: Connect the power adapter that came with the VoIP Gateway. Using a power supply with a

different voltage rating will damage this product. Make sure to observe the proper power requirements. The

power requirement is DC12 volts/0.6 A.

POWER Switch: Power on/off the VoIP Gateway.

WAN Port: Connect to Broadband devices, such as a ADSL or Cable modem.

LAN Port: Connect to Ethernet network devices, such as a PC, hub, switch, or router. Depending on the

connection, you may need a cross over cable or a strait through cable.

RESET: The RESET button will set the VoIP Gateway to its factory default setting and reset the VoIP

Gateway. You may need to place the VoIP Gateway into its factory defaults if the configuration is changed,

you loose the ability to enter the VoIP Gateway via the web interface, or following a software upgrade, and

you loose the ability to enter the VoIP Gateway. To reset the VoIP Gateway, simply press the reset button for

more than 10 seconds. The VoIP Gateway will be reset to its factory defaults and after about 30 seconds the

VoIP Gateway will become operational again.

PHONE Jack: Connect a standard telephone handset to the VoIP Gateway phone jack using a telephone

cable.

2.2 LED Description

PWR LED: The LED stays lighted to indicate the system is power on properly.

SIP LED: This LED is lighted when the VoIP Gateway is REGISTERED successfully to the SIP Server.

ETH LED: The LED is lighted when a connection is established to WAN/LAN port and flashes when

WAN/LAN port is sending/receiving data.

3. Installing VoIP Gateway

3.1 Hardware Installation

1. Locate an optimum location for the VoIP Gateway.

2. For connections to all interfaces, refer to figure below.

3. Connect the AC Power Adapter. Depending upon the type of network, you may want to put the power

supply on an uninterruptible supply. Only use the power adapter supplied with the VoIP Gateway. A different

adapter may damage the product.

Now that the hardware installation is complete, proceed to reset Chapters to set up VoIP Gateway.

3.2 Basic VoIP Configuration

3.2.1 Access to the web configuration of VoIP Gateway

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://10.0.0.2 in the address bar.

3. Entry of the username and password will be prompted. Enter the default login User Name and Password:

The default login User Name of the administrator is admin, and the default login Password is

administrator.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

Step 2:

Now you could configure the VoIP Gateway in detail.

3.2.2 VoIP Configuration

Step 1:

Click " Configuration -> VoIP -> SIP Service Provider "

Step 2:

Click On ratio in Active, enter the information of "Domain Server / Proxy Server / OutboundProxy /

Display Name / User Name / Register Name / Register Password " , which was provided by your

VoIP Service Provider and then click "Submit ".

Step 3:

You have to save and reboot the SIP VoIP Gateway to effect those changes.

Step 4:

Click " Configuration -> Save Settings/Reboot " and then click " Save & Reboot " button.

Step 5:

System will reboot automatically to effect those changes and please wait for a moment while rebooting....

3.

2.3 WAN Configuration

3.2.3.1 Static IP Configuration

Step 1:

Click " WAN -> Fixed IP " and then enter the " IP Address / Subnet Mask / Gateway / DNS Server1

/ DNS Server2 " and then click " Submit "

Step 2:

You have to save and reboot the SIP VoIP Gateway to effect those changes.

Step 3:

Click

" Configuration -> Save Settings/Reboot " and then click " Save & Reboot " button.

Step 4:

System will reboot automatically to effect those changes and please wait for a moment while rebooting....

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the VoIP Gateway is

REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure

configuration again.

3.2.3.2 DHCP Client Mode Configuration

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://10.0.0.2 in the address bar.

3. Entry of the username and password will be prompted. Enter the default login User Name and Password:

The default login User Name of the administrator is admin, and the default login Password is

administrator.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

Step 2:

Click " WAN -> DHCP client " and then click " Submit "

Step 3:

You have to save and reboot the SIP VoIP Gateway to effect those changes.

Step 4:

Click " Configuration -> Save Settings/Reboot " and then click " Save & Reboot " button.

Step 5:

System will reboot automatically to effect those changes and please wait for a moment while rebooting....

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the VoIP Gateway is

REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure

configuration again.

3.

2.3.3 PPPoE Client Mode Configuration

Step 1:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) http://10.0.0.2 in the address bar.

3. Entry of the username and password will be prompted. Enter the default login User Name and Password:

The default login User Name of the administrator is admin, and the default login Password is

administrator.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

Step 2:

Click " WAN -> PPPoE ", enter the " User Name and Password " which was given by Telecom or by

your Internet Service Provider (ISP) and then click " Submit "

Step 3:

You have to save and reboot the SIP VoIP Gateway to effect those changes.

Step 4:

Click " Configuration -> Save Settings/Reboot " and then click " Save & Reboot " button.

Step 5:

System will reboot automatically to effect those changes and please wait for a moment while rebooting....

Please check the SIP LED is lighted or not. If the SIP LED is lighted, the VoIP Gateway is

REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure

configuration again.

4.

Advanced VoIP Configuration

The configuration of VoIP Gateway is web based. The page of VoIP Gateway Configuration can be

reached as follows:

1. Launch the Web browser (Internet Explorer, Netscape, etc.).

2. Enter the LAN port default IP address (default gateway) HYPERLINK "http://10.0.0.2/"http://10.0.0.2/

in the address bar.

3. Username and password will be prompted. Enter the default login User Name and Password:

The default login User Name for administrator is admin, and the default login Password is administrator.

Remember my password checkbox: By default, this box is not checked. Users can check this box so that

Internet Explorer will remember the User name and Password for future logins. It is recommended to leave

this box unchecked for security purposes.

4. On the router Home Page, click the VoIP link on the left frame to view the VoIP Gateway

Configuration page.

In general, configuration changes via web interface will be active only upon clicking Save & Reboot button

on the Save Savings / Reboot page.

Note: Certain Voice Parameters do not require a Save & Reboot to be active. These Voice Parameters will

take effect on the next voice call after the Voice Parameter is entered and submitted. 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.

4.1 Status Page

4.1.1 System Information Page

This page illustrates the system related information.

4.1.2 Network Status Page

You can check the current Network setting in this page.

4.1.3 VoIP Status Page

The page shows current status of VoIP SIP Service provider.

4.2 Configuration Page

4.2.1 WAN Configuration Page

You can configure the WAN settings in this page.

4.2.1.1 The TCP/IP Configuration item defines the LAN port’s network environment. You may refer to

your current network environment to configure the VoIP Gateway properly.

4.2.1.2 The PPPoE Configuration item defines the PPPoE Username and Password. If you have the

PPPoE account from your Service Provider, please insert Username and Password correctly.

4.2.1.3 The Bridge Item defines the VoIP Gateway Bridge mode Enable/Disable. If you set the Bridge On,

then the two Fast Ethernet ports will be transparent.

4.2.1.4 When you complete the setting, please click the Submit button.

4.2.2

LAN Configuration Page

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

4.2.3 VoIP Gateway Configuration Page

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

The VoIP Gateway Configuration page is divided into three general categories: SIP Setting, Phone

Book, Phone Setting, and Others.

4.2.3.

1 SIP Setting Configuration

In SIP Settings you can setup the Service Domain, Port Settings, Codec Settings, RTP Setting, RPort Setting

and Other Settings. If the VoIP service is provided by ISP, you need to setup the related information correctly

then you can register to the SIP Proxy Server correctly.

4.2.3.1.1 SIP Service Provider

In Service Domain Function insert the account and the related information received by your ISP provider.

You can register three SIP accounts in the VoIP Gateway. You can dial your friends' VoIP phone enabling the

first SIP account and receive the call from these three SIP accounts.

SIP Service Provider

Active

First you need click On to enable the Service Domain, then you can input

the following items:

Domain Server

For example, in [email protected], the domain is “domain.com”. Provided

by your VoIP Service Provider.

Proxy Server

If your VoIP service provider has an proxy address and requires that you

provide the address to VoIP Gateway. For the address enter a domain name

(for example, domain.com) or an IP address (for example,

123.456.789.012).

Outbound Proxy

If your VoIP service provider has an outbound proxy address and

requires that you provide the address to VoIP Gateway. For the address

enter a domain name (for example, domain.com) or an IP address (for

example, 123.456.789.012).

Display Name

This name is displayed in the VoIP Gateway display.

Other parties will see this name they are when connected to you.

User Name

Typically the account number for the SIP account.

For example, in [email protected], the user name is “test”.

Provided by your VoIP Service Provider.

Register Name

May not be required.

If it is required, it will be provided by your VoIP Service Provider.

Register Password

Provided by the VoIP Service Provider.

Subscribe for MWI

When set to On a Subscribe for Message Waiting Indication will be

sent periodically.

Register Status

You can see the Register Status in the Status item. If the item shows

“Registered”, then your VoIP Gateway is registered to the ISP, you can

make a phone call directly.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.3.1.2 Port Setting

You can setup the SIP and RTP port number in this page. Each ISP provider will have different SIP/RTP

port setting, please refer to the ISP to setup the port number correctly. When you complete the setting, please

click the Submit button.

4.2.3.1.3 Codec Settings

You can setup the Codec priority, RTP packet length, and VAD(Voice Activity Detection) function

in this page.

Follow the suggestion of your ISP to setup these items. When you complete the setting, please click the

Submit button.

4.2.3.1.4 Codec ID Setting

You can set the value of Codec ID in this page.

4.2.3.1.5 DT MF Setting

You can setup the Out-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 complete 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.

4.2.3.1.6 RPort Function

You can setup the RPort Enable/Disable in this page. To change this setting, please follow your VoIP Service

Provider’s information. When you complete the setting, please click the Submit button.

4.2.3.1.7 QoS

You can setup the Hold by RFC, Voice/SIP QoS, SIP expire time and Use DNS SRV in this page. To

change these settings please follow your ISP information. When you complete the setting, please click the

Submit button.

The QoS sets 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 other Internet devices.

4.2.3.2 Phone Book Configuration

4.2.3.2.1 The Phone Book contains Speed Dial Settings. You can setup the Speed Dial number. If you want

to use Speed Dial, just dial the speed dial number then press “#”.

4.2.3.2.2 In the Phone Book setting function you can add/delete Speed Dial number. You can insert 140

entries maximum in the Speed Dial list.

4.2.3.2.2.1 To add a phone number in the Speed Dial list, insert the position, the name (Speed Dial Number),

and the phone number (by URL type). When you complete a new phone list, just click the “Add Phone”

button.

4.2.3.2.2.2 If you want to delete a phone number, select the phone number you want to delete then click the

“Delete Selected” button.

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

Phone Book Page

Book Page

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

Phone

Show the phone number by sequence. There are total 140 phone numbers

from Phone 0 to Phone 139 can be set

Name

Enter the Name

Number

Enter the Speed Dial Number

URL

Display the URL that you configured

Select

Select the item of the phone number

Delete Selected

[Button]

Delete selected item

Delete All [Button]

Delete all items

Reset [Button]

Reset selected item

Add New Phone

Position

Enter the phone number from 0 to 139

Name

Enter the Name

Number

Enter the Speed Dial Number

URL

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

Gateway

Add Phone
[Button]

Add the new Phone which you configured

Reset [Button]

Reset configured items

Examples

Example 1: Position: 0, Name: IPtel User test, Number: 000, URL: HYPERLINK

"mailto:[email protected]" [email protected]

When the user dials the Number 000, he will call the VoIP User test who is registered to the SIP Server

iptel.org.

Please note that you need also to register to the SIP Server iptel.org. If you register to different SIP Server,

please make sure that the SIP Server allows you to call iptel.org.

Example 2: Position: 1, Name: IP Dialing #1, Number: 001, URL: 192.168.10.32

When the user dials the Number 001, he will call the VoIP Device whose WAN IP Address is 192.168.10.32.

Example 3: Position: 2, Name: IP Dialing #2, Number: 002, URL: 192.168.10.132:5062

When the user dials the Number 002, he will call the VoIP Device whose WAN IP Address is

192.168.10.132 with the port 5062.

Example 4: Position: 3, Name: VoIP User 88888888, Number: 003, URL: 88888888

When the user dials the Number 003, he will call the VoIP User whose phone number is 88888888.

Example 5: Position: 4, Name: VoIP User voipuser, Number: 004, URL: voipuser

When the user dials the Number 004, he will call the VoIP User whose phone number is voipuser.

Example 6: Position: 5, Name: VoIP Out #1, Number: 005, URL: 000019998887777

When the user dials the Number 005, he will call the PSTN phone number 000019998887777 by VoIP

OUT.

Important notice: make sure that your VoIP Service Provider supports the VoIP OUT.

If your VoIP Service Provider supports the VoIP OUT, please follow the instructions of your

VoIP Service Provide to dial the PSTN phone number by VoIP OUT.

For example: as suggested by the VoIP Server Provider, dial the recommended dialing

sequence: 00 + country code + telephone number (e.g. 00 1 999 888 7777).

Example 7: When dialing a VoIP Phone Number that isn’t configured in the Number list, it will dial out the

VoIP Phone Number.

4.2.3.3 Phone Setting

The Phone Setting contains the following functions: 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.

4.2.3.3.1 Call Forward function

In this page you can setup the phone number you want to forward. There are three type of Forward mode.

You can choose All Forward, Busy Forward, and No Answer Forward by click the icon.

All Forward

All incoming call will be forwarded to the

URL/number you configured.

Busy Forward

If you are on the phone, the new incoming call will be

forwarded to the URL/number you configured.

No Answer Forward

If you can not answer the phone after a specific ring

you configured, the incoming call will be forwarded

to the URL/number you configured.

Off

Disable call forward.

IP

Enable call forward for URL/number.

PSTN (Optional)

Enable call forward for PSTN phone number. Only

the for 1 FXO +1 FXS

All Fwd No.

The URL/number you configured will be forwarded

to for All Forward

Busy Fwd No.

The URL/number you configured will be forwarded

to for Busy Forward

No Answer Fwd No.

The URL/number you configured will be forwarded

to for No Answer Forward

Name

Display the name of URL/number you configured

URL

Enter the URL, VoIP Phone Number, Remote WAN

IP Address of VoIP Gateway to which you want the

call is to be forwarded to.

No Answer Fwd Time Out

You can set the Time Out time for system to start the

call forwarding to the number you configured for No

Answer Forward

Submit Button

When you complete the setting, please click the

Submit button.

Reset Button

You can reset the configured parameters before you

submit.

Back Button

Go back to the previous web page

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

All incoming calls will be forwarded to the VoIP phone number 7777.

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

All incoming calls will be forwarded 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 calls will be forwarded 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 be forwarded to the VoIP phone number 7777.

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 be forwarded 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 be forwarded 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 be forwarded 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 be forwarded to the PSTN phone number

88888888.

4.2.3.3.2 Volume Setting function

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

When you complete the setting, please click the Submit button.

4.2.3.3.2.1 Handset Volume sets the volume of the earphone of your handset.

4.2.3.3.2.2 PSTN-Out Volume sets the PSTN volume of the microphone of your handset, sent out to the

other side's earphone of handset.

4.2.3.3.2.3 Handset Gain sets the volume of the microphone of your handset, sent out to the other side's

earphone of handset.

4.2.3.3.2.4 PSTN-In Gain sets the PSTN volume of the earphone of your handset.

4.2.3.3.2.5 When you complete the setting, please click the Submit button.

4.2.3.3.3 DND Setting function

In this page you can set the do not disturb period of your phone.

DND Always

Default is Off (disable). When it was On (enable). All incoming call will

be blocked and the caller will hear the busy tone any time when place a call

until disable this feature.

DNS Period

Default is Off (disable). When it was On (enable). All incoming call will

be blocked and the caller will hear the busy tone any time when place a call

during the time period until disable this feature. If the “From” time is large

than the “To” time, the Block time will from Day 1 to Day 2.

From

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

To

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

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.3.3.4 Caller ID function

You can set the device to show Caller ID in your PSTN Phone or IP Phone. There are four selections for Caller

ID. The setting of the Caller ID function for FSK or DTMF depends on your phone.

Single Caller ID

Default is Off (disable). When it was Yes (enable), It’ll detect the Singel

Caller ID.

CID Without Time

Default is Off (disable). When it was Yes (enable), It’ll detect the Caller

ID without time.

CID Type 2

Default is Off (disable). When it was Yes (enable), It’ll detect the Caller

ID Type 2.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.3.3.5 Dial Plan function

Number to add or replace before dial the phone number.

Drop Prefix

Default is NO (Add the Prefix). When it was Yes (Drop the Prefix),

It’ll drop the prefix.

NO (Add the Prefix): When it meets the rule which you configured, it’ll

add the prefix. Maximum input digits are 7.

Yes (Drop the Prefix): When it meets the rule which you configured, it’ll

drop the prefix and replace the number which you configured. Maximum

input digits are 31.

Replace rule1
Replace rule2

There are total 4 replace rules for use.

Replace rule3

Replace rule4

+: or

xxx: Define the length of digits.

Dial now

If the numbers which you dialed met this rule, it will dial out with its dial

plan immediately.

Be noted that the first digit cannot be 0 due to 0 in the first digit is to ignore

this rule. If you set the rule 0xxxxx and this rule is invalid due to the first

digit is 0.

Auto Dial Time

Default is 5 (Seconds). How long the phone number will be dialed out

after finishing dialing the digits.

Use # as send key

Default is Yes. When it was No, It’ll wait for the setting of Auto Dial Time

and then dial out after dialing the phone numbers.

Use * for IP dialing

Default is Yes. When it was No, the * key will not be as . for IP Dialing.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

Symbol explanation:

x or X

0,1,2,3,4,5,6,7,8,9

+

or

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

When the number 8613 is dialed and the prefix 002is added, the real phone number [002+8613+xxx] will

be dialed out.

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

00286135555 will be dialed out.

When the number 8662 is dialed and the prefix 002 is added, the real phone number [002+8662+xxx]

will be dialed out.

For example, when you dial the number 86625555, 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 is dialed, the digits 002 will be replaced with 006 and the whole digits [006+xxx]

will be dialed out.

For example, when you dial the number 0025555 and the digits 002 is replaced with 006, then the real

phone number 0065555 will be dialed out.

When the number 003 is dialed, the digits 003 is replaced with 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 with 006, then the

real phone number 0065555 will be dialed out.

When the number 004 is dialed, the digits 004 will be replaced with 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 with 006, then the

real phone number 0065555 will be dialed out.

When the number 005 is dialed, the digits 005 will be replaced with 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 with 006, then real

phone number digits 0065555 will be dialed out.

When the number 007 is dialed, the digits 007 will be replaced with 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 with 006, then the

real phone number 0065555 will be dialed out.

When the number 009 is dialed, the digits 009 will be replaced with 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 with 006, then the

real phone number 0065555 will be dialed out.

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

When the number 12 is dialed, the prefix 009 is 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, 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 is dialed and the prefix 007 is added, the whole digits [007+5xxx] will be dialed

out. 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 is added, the real phone number 0075999

will be dialed out.

When the number 35xx is dialed and the prefix 007 is added, the whole digits [007+35xx] will be dialed

out. Note that the range of xx is from 00 to 99.

For example, when you dial the number 3500 and the prefix 007 is added, the real phone number

0073500 will be dialed out.

For example, when you dial the number 3599 and the prefix 007 is added, the real phone number 0073599

will be dialed out.

When the number 21xx is dialed, the prefix 007 is added and the whole digits [007+21xx] will be dialed

out. Note that the range of xx is from 00 to 99.

For example, when you dial the number 2100, the prefix 007 is added and the real phone number 0072100

will be dialed out.

For example, when you dial the number 2199 and the prefix 007 is added, the real phone number 0072199

will be dialed out.

When the number 534 is dialed, the prefix 007 will not be added and the real phone number 534 will be

dialed out because the above mentioned rule is not matched.

When the number 358822 is dialed, the prefix 007 will not be added and the real phone number 358822

will be dialed out because the above mentioned rule is not matched.

Example 5: Dial Now: xx

When the two digits in the range from 00 to 99 have been dialed, they will be dial out immediately.

Auto Dial Time function

It is when you insert the phone number by the keypad and you don’t need to press “#”. After time out the

system will dial directly.

Auto Dial Time function

The * key will not be as “.” in IP Dialing. If you want to dial the IP Dialing, you need to know the WAN IP

Address of the remote VoIP Devices. For example if the WAN IP Address of Remote VoIP Device is

222.222.222.222, then you need to dial 222*222*222*222# to make a IP Dialing.

4.2.3.3.6 Flash Time Settings function

When you use the PSTN Phone and you need to press the Hook to do the Flash(Switch to the other phone

line or HOLD). This function sets the time you must press the Hook to activate the Flash function.

4.2.3.3.7 Call Waiting Settings

In this page you can enable/disable the call waiting setting.

If a new call is coming while you are talking, you can press the Flash button to switch to the new call.

Pressing the Flash button you switch between the two calls.

To end the first call, hang up the phone. Then the phone will ring, please pick it up to answer the second call.

Hang up again to end the call.

4.2.3.3.8 T.38 (FAX) Setting

In this page you can enable/disable the FAX function.

T.38 (FAX)

Default is Off (Disabled). When it is On

(Enabled), it enables the T.38 Fax function.

T.38 Port/ T.38 Port of Phone1

Default is 60000. (Only one port at a time is

supported)

T.38 Port of Phone2

Default is 60100. (Only one port at a time is

supported)

Submit Button

When you complete the setting, please click the

Submit button.

Reset Button

You can reset the configured parameters before you

submit

Back Button

Go back to the previous web page

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 in its entirety in-band over a voice call (over RTP). The VoIP Gateway is aware that the call in

progress is a fax call and not a 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, if necessary, turn off echo cancellation (EC) and voice

activity detection (VAD) and fix 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. Annexe D describes the system level

requirements and procedures for establishing fax calls between two SIP based endpoints. In this mode, the

VoIP Gateway will establish a normal voice call and switch to fax based on the detection of Fax tones from

the PTM. It will then 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 IFP packets can be sent over TCP or

UDP (VoIP Gateway supports only UDP). On call disconnect, SIP signaling is used to end the call.

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). The VoIP Gateway supports both kinds of

behaviors.

4.2.3.3.

9 Hot line Settings

Provide the Hot Line function.

It'll dial the configured URL, VoIP Phone Number or the Remote WAN IP Address of VoIP Gateway

automatically every time you pick up the phone.

Use Hot Line

Default is Disable. When it is Enable, it

enables the Hot Line function.

Hot Line Number

Enter the URL, VoIP Phone Number, Remote

WAN IP Address of VoIP Gateway you want to

use for Hot Line.

Submit Button

When you finished the setting, please click the

Submit button.

Reset Button

You can reset the configured parameters before

you submit

Back Button

Go back to the previous web page

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

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

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

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

Example 3: Use Hot Line: Enable, Hot line number: 192.168.10.63

Every time you pick up the phone, it will dial the WAN IP Address 192.168.10.63 of Remote VoIP Gateway

automatically.

4.2.3.4 Others

This section contains Auto Configuration Settings, FXO & FXS Impedence Setting, MAC Clone

Settings and Advanced Settings functions.

4.2.3.4.1 Auto Configuration Settings

In this page you can enable/disable the auto configuration/provisioning settings.

The VoIP Gateway provides secure provisioning and remote upgrade. Provisioning is achieved through

configuration profiles transferred to the device via TFTP, HTTP or FTP. The VoIP Gateway can be configured

to update its VoIP Configuration from a remote profile at power up or reboot.

Auto Configuration

Default is Off(Disable). When it was Enable, there are 3 types of Auto

Configuration: TFTP, FTP and HTTP.

TFTP Server

Enter IP or Domain Name of TFTP Server.

HTTP Server

Enter IP or Domain Name of HTTP Server.

HTTP Path

Enter File Path where the provisioning file is.

FTP Server

Enter IP or Domain Name of FTP Server.

FTP Username

Enter Username which provided by FTP Server.

FTP Password

Enter Password which provided by FTP Server.

File Path

Enter File Path where the provisioning file is.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

Example 1: Auto Configuration for HTTP Server

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

Every time you power on the VoIP Gateway, 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 you power on the VoIP Gateway, 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 you power on the VoIP Gateway, it will update its VoIP configuration to the latest one from Auto

Provisioning Server (FTP Server) automatically.

4.2.3.4.2 FXO & FXS Impedence Setting

In this page you can select the FXO & FXS Impedence Setting for different countries.

FXO Port

Default is USA. You could select the FXO Impedence Setting for different country

here.

FXS Port

Default is USA. You could select the FXS Impedence Setting for different country

here.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.3.4.3 STUN Setting

In this page you can Enable/Disable the STUN and configure the STUN Server IP address.

This function helps your VoIP Gateway working properly behind NAT. To change these settings please follow

your VoIP Service Provider’s information. When you complete the setting, please click the Submit button.

STUN

Default is Off (disable). When it was On (enable). It enables STUN

(Simple Transversal of UDP through NAT) if the VoIP Gateway 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

Enter the IP address or Domain Name of the STUN Server. The default is

stun.xten.com. This field is applicable only if USE STUN is selected as the

NAT traversal technique.

STUN Port

Enter the port number on which the STUN server listens for requests from

the STUN Client on VoIP Gateway. The range is 1024 to 65535. The default

is 3478. This field is applicable only if USE STUN is selected as the NAT

traversal technique.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.3.4.4 MAC Clone Settings

Some ISPs do not want you to have a home network and have a DSL/Cable modem that allows only 1 MAC to

talk on the internet. If you change the network cards, you have to call them up to change the MAC. The VoIP

Gateway can clone the computer's MAC that was originally set up for such an ISP.

MAC Clone

Default is Off (disabled). When it is On

(enabled), the VoIP Gateway clones the

computer's MAC that was originally set up for

that ISP.

Submit Button

When you complete the setting, please click the

Submit button.

Reset Button

You can reset the configured parameters before

you submit

Back Button

Go back to the previous web page

4.2.3.4.5 Tones settings

In this page you can configure your tones settings.

4.2.

3.4.6 Advanced Settings

In this page you can change advanced setting.

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

"Calling Party" has hung up. The CPC signal tells the phone equipment that the outside party has hung-up,

so it can stop recording to an answering machine or voice mail, drop the call off hold, or just release a line

that might be used for dictation or announcements.

ICMP Not Echo

Default is Off (disable). When it was On (enable). The VoIP Gateway

will not echo the ICMP request.

Send Anonymous

CID

The Anonymous Caller ID to display when you make a call to others VoIP

Gateways.

Billing Signal

Default is Off (disable). When it was On (enable). Polarity Reversal is

enabled to inform the charge/billing system (Polarity Reversal, Tone_12K,

Tone_16K). Support FXS Port only

CPC Delay

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

you configuration. Support FXS Port only

CPC Duration

When VoIP Gateway is the called party, CPC duration is the "voltage drop"

duration, before it plays dial tone again. Support FXS Port only

Send Flash event

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

SIP Info.

SIP Encrypt

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

WALKERSUN1, WALKERSUN2, CSF1, CSF2 and GX.

PPPoE retry period

(*)

Default is 5 seconds. The range is 5 to 255. When PPPoE failed to connect to

ISP, it will wait for the period which you configured to redial.

System Log Server

To upload the system log on the specified Server

System Log Type

Default is None. There are 7 types: Call Statistics, General Debug, Call

Statistics + General Debug, SIP Debug, Call Statistics + SIP

Debug, General Debug + SIP Debug, All.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

Example 1: ICMP Not Echo: Yes

The ICMP will not echo no matter you request from LAN side or WAN side.

Example 2: Send Anonymous CID: Yes

Every time you make a call to others VoIP Gateways, it willl send the Anonymous as Caller ID out

automatically.

Example 3: Management from WAN: Yes

You can remote manage from the WAN IP Address of the VoIP Gateway.

Example 4:

Send Flash event: DTMF EVENT

It will send the DTMF EVENT as Flash event.

Send Flash event: SIP INFO

It will send the SIP INFO as Flash event.

4.2.4

DDNS Configuration Page

In this page you can configure the DDNS setting. You must have the DDNS account and insert the

information properly. You can have a DDNS account with a public IP address so that others can call you via

the DDNS account. But now most of the VoIP applications are working with a SIP Proxy Server. When you

complete the setting, please click the Submit button.

Example 1:

Configure the WAN to PPPoE Client and make sure you got the WAN IP Address (Public IP Address).

Configure the Host Name, User Name, Password, and E-mail Address.

If every parameter is correctly configured, you can visit the home page of the VoIP Gateway entering the

DDNS Host Name as follow.

4.2.5 VLAN Settings Page

In this page you can set the VLAN settings.

VLAN Packets

Default is Off(Disable). When it was On(Enable), It’ll enable to receive

VLAN Packets function.

VID (802.1Q/TAG)

Default is 136. 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.

User Priority

(802.1P)

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).

CFI

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.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Back Button

Go back to the previous web page

4.2.6 Virtual Server Page

In this page you can configure your demilitarized zone setting.

4.2.7

Virtual Server Page

Virtual Servers are used for port forwarding from the WAN to LAN networks. The Virtual Server

Configuration page allows you to set the configuration of the Virtual Server. All UDP/TCP ports are protected

from intrusion. If any specific local PCs need to be mapped to the UDP/TCP port on WAN side, please insert

the mappings here.

There can be up to 24 different Virtual Server Configurations.

Virtual Server Page

Virtual Server Page

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

Num

Show the number by sequence. There are total 24 numbers from Phone 0 to

Phone 23 can be set

This is the number corresponding to the Virtual Server configuration.

Enable

Default is Disable. When it was Enable, It’ll enable the Virtual Server

Protocol

Select TCP or UDP.

In Port (Internal

Port)

Display the Internal Port that you configured

Ex Port (External

Port)

Display the External Port that you configured

Server IP

Display the private network IP address for the particular server.

Select

Select the item of the Virtual Server

Enable Selected

Enable selected item

[Button]

Delete Selected

[Button]

Delete selected item

Delete All [Button]

Delete all items

Reset [Button]

Reset selected item

Add Virtual Server

Num

Enter the number corresponding to the Virtual Server configuration.

Server IP

Enter the private network IP address for the particular server.

Protocol

Select TCP or UDP.

Internal Port

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.

External Port

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

Add Server
[Button]

Add the new Server which you configured

Reset [Button]

Reset configured items

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 VoIP Gateway and then the VoIP

Gateway will re-direct it to your LAN IP 10.0.0.150.

Table 4

-3. 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

4.2.8 PPTP Settings Page

A VPN is a private network of computers that uses the public Internet to connect some nodes. Because the

Internet is essentially an open network, the Point-to-Point Tunneling Protocol (PPTP) is used to ensure that

messages transmitted from one VPN node to another are secure. With PPTP, users can dial in to their

corporate network via Internet.

PPTP Settings Page

PPTP

Default is Off. When it was On, It’ll enable the PPTP client.

PPTP Server

Enter the IP Address of PPTP Server.

PPTP Username

Enter the Username of PPTP client.

PPTP Password

Enter the Pasword of PPTP client.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

4.2.9 SNTP Settings Page

You can setup the primary and second SNTP Server IP Address, to get the date/time information. Also you

can base on your location to set the Time Zone, and how long need to synchronize again. When you complete

the setting, please click the Submit button.

If synchronization is enabled, your VoIP Gateway 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.

If time synchronization fails, it might be due to:

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 Internet time server is too busy or is temporarily unavailable. If this is the case, try synchronizing your

clock later, or update it manually by powering off and then on the VoIP Gateway. You can also try using a

different time server.

The time shown on your VoIP Gateway is too different from the current time on the Internet time server.

Internet time servers might not synchronize your clock if your VoIP Gateway's time is off by more than 15

hours.

4.2.10 Alarm Settings Page

It provides the alarm function.

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

Alarm

Default is OFF (Disabled). When it is ON

(Enabled), it will enable the Alarm function.

Alarm Time

Default is 0:0 (hh:mm). Set the Alarm Time.

(24 hours format, hh:mm)

Current time

It’s the current time of the VoIP Gateway.

Submit Button

When you complete the setting, please click the

Submit button.

Reset Button

You can reset the configured parameters before

you submit

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

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

Example 2: Alarm: ON, Alarm Time: 23:31(hh:mm)

The alarm will sound when it reached the current time 23:31.

4.2.11 System Authority Page

In the System Authority you can change your login name and password.

4.2.12 Save Settings/Reboot Page

In Save Settings/Reboot you can save the changes you have done or reboot only. If you want to use new

setting in the VoIP Gateway, you have to click the Save & Reboot button. After you click the Save &

Reboot button, the VoIP Gateway will automatically restart and the new setting will be effective.

If you want to reboot the VoIP Gateway, you have to click the Reboot Only button. After you click the

Reboot Only button, the VoIP Gateway will automatically restart.

4.3

System Page

4.3.1 Reset factory default Page

In Reset to Factory Default setting you can restore the VoIP Gateway to factory default. Just click the Restore

button, the VoIP Gateway will restore to default and automatically restart.

4.3.2 Firmware Update Page

In Update you can update the VoIP Gateway’s firmware to the new one or do the factory reset to let the VoIP

Gateway back to default setting.

Click the “Browse” button in the right side of the File Location or type the correct path and the filename in

File Location blank and then click the Update button.

4.3.3

Auto Update Page

To update the firmware, power on the VoIP Gateway or Scheduling.

Update via

Default is OFF (Disable). When it was TFTP/FTP/HTTP(Enable), it’ll

enable the auto update function and request from the TFTP/FTP/HTTP

Server.

TFTP Server

Enter IP or Domain Name of TFTP Server.

HTTP Server

Enter IP or Domain Name of HTTP Server.

HTTP Path

Enter File Path where the file is.

FTP Server

Enter IP or Domain Name of FTP Server.

FTP Username

Enter Username which provided by FTP Server.

FTP Password

Enter Password which provided by FTP Server.

File Path

Enter File Path where the file is.

Check new

firmware

Power ON: It’ll check if there is a new firmware on the TFTP/FTP/HTTP

Server by powering on the VoIP Gateway.

Scheduling: It’ll check if there is a new firmware on the TFTP/FTP/HTTP

Server

by scheduling.

Scheduling (Date)

Default is 14. It’ll check if there is a new firmware on the TFTP/FTP/HTTP

Server periodically. The range of the Scheduling Date is 1 - 30.

Scheduling (Time)

Default is AM 00:00- 05:59. It’ll check if there is new firmware on the

TFTP/FTP/HTTP Server periodically. There are four Scheduling Time:

AM 00:00- 05:59, AM 06:00- 11:59, PM 12:00- 17:59, PM 18:00-

23:59

Automatic Update

Notify only: When there is a newer firmware, it will only notify by “BEEP

BEEP BEEP” you when you pick up the phone.

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

the firmware automatically.

Firmware File

Prefix

The file prefix of the firmware

Next update time

It’s the next update or check time.

Submit Button

When you finished the setting, please click the Submit button.

Reset Button

You can reset the configured parameters before you submit

Example 1: HTTP - Firmware update by notification when powered on

Auto Update Settings

Update via: HTTP

HTTP Server: 192.168.10.100

HTTP Path: /

Check new Firmware: Power ON

Automatic Update: Notify only

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

Every time you power on the VoIP Gateway, it will notify you with “BEEP BEEP BEEP” that there is an up

to date firmware available on HTTP Server after you pick up the phone; you can update the firmware

manually.

Create the Auto Update files on HTTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the

TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Power on the VoIP Gateway and it will check if there is any firmware update on the Server. If a newer

firmware is avilable, it will only notify you with “BEEP BEEP BEEP” after you pick up the phone.

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

Pick up the phone again and then press #160# , then hang up the phone to have your VoIP Gateway

firmware upgraded immediately.

It takes about 3 minutes for updating the new firmware and the SIP LED starts blinking while updating the

firmware.

Once the SIP LED stop blinking, please power off and then power on the VoIP Gateway to active the new

firmware.

Example 2: TFTP - Firmware update by notification when powered on

Auto Update Settings

Update via: TFTP

TFTP Server: 192.168.10.100

Check new Firmware: Power ON

Automatic Update: Notify only

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

Every time you power on the VoIP Gateway, it will notify you with “BEEP BEEP BEEP” there is an up to

date firmware available on TFTP Server after you pick up the phone; you can update the firmware

manually.

Create the Auto Update files on TFTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the

TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time you power on the VoIP Gateway, it will check if there is an up to date firmware available on TFTP

Server and update the firmware manually. When there is a newer firmware, it will only notify you with

“BEEP BEEP BEEP” after you pick up the phone.

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

Pick up the phone, press #160# and then hang up the phone to have VoIP Gateway firmware updated

immediately.

It takes about 3 minutes to update the new firmware and the SIP LED starts blinking while updating the

firmware.

When the SIP LED stops blinking, VoIP Gateway will reboot itself to active the new firmware.

Example 3: FTP - Firmware update by notification when powered on

Auto Update Settings

Update via: FTP

FTP Server: 192.168.10.100

FTP Username: 1234

FTP Password: 1234

File Path: /

Check new Firmware: Power ON

Automatic Update: Notify only

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

Every time you power on the VoIP Gateway, it will notify you with “BEEP BEEP BEEP” an up to date

firmware is available on FTP Server after you pick up the phone; you can update the firmware manually.

Create the Auto Update files on FTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time you power on the VoIP Gateway, it wil check if there is an up to date firmware available on FTP

Server and update the firmware manually. If there is a newer firmware, it will only notify you with “BEEP

BEEP BEEP” after you pick up the phone.

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

Pick up the phone and then press #160#, then hang up the phone to have VoIP Gateway firmware updated

immediately.

It takes about 3 minutes to update the new firmware, the SIP LED starts blinking while updating the

firmware.

Once the SIP LED stops blinking, the VoIP Gateway will reboot itself to active the new firmware.

Example 3: FTP - Firmware update by notification when reached the Scheduling Date and

Time

Auto Update Settings

Update via: FTP

FTP Server: 192.168.10.100

FTP Username: 1234

FTP Password: 1234

File Path: /

Check new Firmware: Scheduling

Automatic Update: Notify only

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

It will update its VoIP firmware to the latest version from FTP Server automatically when it reaches the

Scheduling Date and Scheduling Time (Next update time).

Create the Auto Update files on FTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time the VoIP Gateway reaches the scheduling date and time, it will notify you with “BEEP BEEP

BEEP” an up to date firmware is available on FTP Server after you pick up the phone and you can update

the firmware manually.

Be noted:

If the VoIP Gateway is powered off and passed the Next update time, it will not update the firmware after

you power on the VoIP Gateway. It will only update when the VoIP Gateway is powered on and reaches Next

update time.

If you are on the phone having a conversation via VoIP and the Next update time is passing, it will update

the firmware immediately after you hang up the phone.

Example 3: Firmware update by notification when reached the Scheduling Date and Time

Auto Update Settings

Update via: HTTP

HTTP Server: 192.168.10.100

HTTP Path: /

Check new Firmware: Scheduling

Automatic Update: Automatic (Scheduling)

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

It will update its firmware to the latest one from HTTP Server automatically when it reaches the

Scheduling Date and Scheduling Time (Next update time).

Create the Auto Update files on HTTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time the VoIP Gateway reaches the Scheduling date and time, it will check if there is an up to date

firmware available on HTTP Server and update the firmware automatically.

It takes about 3 minutes for updating the new firmware, the SIP LED starts blinking while updating the

firmware.

Once the SIP LED stop blinking, please power off and then power on the VoIP Gateway to active the new

firmware.

Be noted:

If the VoIP Gateway is powered off and passed the Next update time, it will not update the firmware after

you power on the VoIP Gateway. It will only update when the VoIP Gateway is powered on and reaches Next

update time.

If you are on the phone having a conversation via VoIP and the Next update time is passing, it will update

the firmware immediately after you hang up the phone.

Example 4: Firmware update automatically when reached the Scheduling Date and Time

Auto Configuration Settings

Update via: TFTP

TFTP Server: 192.168.10.100

Check new Firmware: Scheduling

Automatic Update: Automatic (Scheduling)

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

It will update its firmware to the latest one from TFTP Server automatically when it reaches the

Scheduling Date and Scheduling Time (Next update time).

Create the Auto Update files on TFTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the

TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time the VoIP Gateway reaches the Scheduling date and time, it will check if there is an up to date

firmware available on TFTP Server and update the firmware automatically.

It takes about 3 minutes to update the new firmware, the SIP LED starts blinking while updating the

firmware.

Once the SIP LED stops blinking, the VoIP Gateway will reboot itself to active the new firmware.

Be noted:

If the VoIP Gateway is powered off and passed the Next update time, it will not update the firmware after

you power on the VoIP Gateway. It will only update when the VoIP Gateway is powered on and reaches Next

update time.

If you are on the phone having a conversation via VoIP and the Next update time is passing, it will update

the firmware immediately after you hang up the phone.

Example 5:

Auto Configuration Settings (Firmware update by Scheduling)

Update via: FTP

Update via: FTP

FTP Server: 192.168.10.100

FTP Username: 1234

FTP Password: 1234

File Path: /

Check new Firmware: Scheduling

Automatic Update: Automatic (Scheduling)

Firmware File Prefix: TA2S

RULE of AUTO UPDATE:

It will update its firmware to the latest one from FTP Server automatically when it reaches the Scheduling

Date and Scheduling Time (Next update time).

Create the Auto Update files on FTP Server:

1. To check the current firmware version of the VoIP Gateway:

a. Telnet 10.0.0.2

b. Enter the login name admin and password administrator.

c. ver

d. You will get the firmware version as follow:

Firmware Version: V701240

2. Create a TA2S_ver.dat due to format of the file is Firmware File Prefix_ ver.dat and edit the

content as follow:

Version: 701250 NAME:TA2S_

3. Change the new firmware voip.gz to TA2S_701250.gz

4. Put the TA2S_701250.gz and TA2S_ver.dat in Server

AUTO UPDATE PROCEDURES:

Every time the VoIP Gateway reaches the Scheduling date and time, it will check if there is an up to date

firmware available on FTP Server and update the firmware automatically.

It takes about 3 minutes to update the new firmware, the SIP LED starts blinking while updating the

firmware.

Once the SIP LED stops blinking, the VoIP Gateway will reboot itself to active the new firmware.

Be noted:

If the VoIP Gateway is powered off and passed the Next update time, it will not update the firmware after

you power on the VoIP Gateway. It will only update when the VoIP Gateway is powered on and reaches Next

update time.

If you are on the phone having a conversation via VoIP and the Next update time is passing, it will update

the firmware immediately after you hang up the phone.

5. IVR Interface for VoIP Gateway

You can use the PSTN phone to configure the VoIP Gateway. Please follow the instruction to configure your

VoIP Gateway.

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

“User-Changeable”

NONDEFAULT 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

#122#

None

IVR will report current in use

VoIP number

Info

Check Network

Mask

#123#

None

IVR will report the WAN Port

network mask

Info

Check Gateway IP

Address

#124#

None

IVR will announce the current

gateway IP address of the VoIP

Gateway

Info

Check Primary

DNS Server Setting

#125#

None

IVR will announce the current

setting in the Primary DNS

field.

Info

Check IP Address

#126#

None

IVR will report the WAN port

IP address

Info

Check Firmware

Version

#128#

None

IVR will announce the version

of the firmware running on the

VoIP Gateway.

6.

How to make a phone call

When your VoIP Gateway is configured properly, you can make a phone call to your friend with the same

Service provider. Please make sure all the cables are connected properly, like PSTN Line cable, Phone cable,

Ethernet cable, Power cable.

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

dialing of the phone number.

6.1 Dial a PSTN Phone call

Default the VoIP Gateway is set in VoIP Phone Call mode. If you want to make a phone PSTN call, you can

press “0*”, dial the phone number and press “#” button to start to dial the phone number.

For example: 0* + phone number + #

6.2 Dial a VoIP Phone call

When your VoIP Gateway is configured properly, you can make a phone call to your friend in the same

Service provider.

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

phone number.

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

6.2.1 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.

6.2.2 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.

6.2.3

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’ll transfer the first call to the second call.

6.2.4 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.

or

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.

6.2.5 Call Hold

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

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

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

7

. Trouble Shooting

7.1 To check what the Internet/WAN access if your own Network is DHCP Client, Static IP or
PPPoE Client

1. Please make sure that you have the Interent/WAN access before changing to the VoIP Gateway, please

check if you could surf the Internet. If you could surf the Internet, you have the Interent/WAN access.

7.1.1 If your Internet/WAN access is the PPPoE client

a. You'll have a shortcut of PPPoE dial up connection on desktop as follow

b. When PPPoE dial up connection connected, there will be an icon of PPPoE dial up connection showed in

notification area as follow.

You could double click on the icon and then click on the Details tab to check the detailed information as

follow. If you could see the Device Tyep is PPPoE, your WAN access is the PPPoE client.

Or click on Start Menu -> Run .. -> enter command -> click OK -> enter ipconfig /all and then press

enter key. The detail IP configuration is showed as follow:

If you could see the PPP adapter as follow, your WAN access is the PPPoE client:

C:\Documents and Settings\ASUS P4P800VM>ipconfig /all

Windows IP Configuration

Host Name . . . . . . . . . . . . : asus-00ef66a32b

Primary Dns Suffix . . . . . . . :

Node Type . . . . . . . . . . . . : Unknown

IP Routing Enabled. . . . . . . . : No

WINS Proxy Enabled. . . . . . . . : No

DNS Suffix Search List. . . . . . : local.lan

Ethernet adapter Local Area Connection:

Connection-specific DNS Suffix . : local.lan

Description . . . . . . . . . . . : Intel(R) PRO/100 VE Network Connection

Physical Address. . . . . . . . . : 00-11-2F-28-29-E5

Dhcp Enabled. . . . . . . . . . . : Yes

Autoconfiguration Enabled . . . . : Yes

IP Address. . . . . . . . . . . . : 10.0.0.6

Subnet Mask . . . . . . . . . . . : 255.255.255.0

Default Gateway . . . . . . . . . : 10.0.0.2

DHCP Server . . . . . . . . . . . : 10.0.0.2

DNS Servers . . . . . . . . . . . : 10.0.0.2

Lease Obtained. . . . . . . . . . : Thursday, August 03, 2006 12:24:31 AM

Lease Expires . . . . . . . . . . : Thursday, August 03, 2006 12:24:31 PM

PPP adapter 1234:

Connection-specific DNS Suffix . :

Description . . . . . . . . . . . : WAN (PPP/SLIP) Interface

Physical Address. . . . . . . . . : 00-53-45-00-00-00

Dhcp Enabled. . . . . . . . . . . : No

IP Address. . . . . . . . . . . . : 192.168.10.204

Subnet Mask . . . . . . . . . . . : 255.255.255.255

Default Gateway . . . . . . . . . : 192.168.10.204

DNS Servers . . . . . . . . . . . : 192.168.10.100

NetBIOS over Tcpip. . . . . . . . : Disabled

For Windows 98/ME user who will see the PPP Adapter as follow:

1 Ethernet adapter :

Description . . . . . . . . : PPP Adapter.

Physical Address. . . . . . : 44-45-53-54-00-00

DHCP Enabled. . . . . . . . : Yes

IP Address. . . . . . . . . : 192.168.10.207

Subnet Mask . . . . . . . . : 255.255.255.0

Default Gateway . . . . . . : 192.168.10.207

DHCP Server . . . . . . . . : 255.255.255.255

Primary WINS Server . . . . :

Secondary WINS Server . . . :

Lease Obtained. . . . . . . :

Lease Expires . . . . . . . :

7.1.2 If your Internet/WAN access is the DHCP client

Click on Start Menu -> Run .. -> enter command -> click OK -> ipconfig /all. The detail IP

configuration is showed as follow:

If the Dhcp Enabled is Yes in Ethernet adapter Local Area Connection, your WAN access is

the DHCP client:

C:\Documents and Settings\ASUS P4P800VM>ipconfig /all

Windows IP Configuration

Host Name . . . . . . . . . . . . : asus-00ef66a32b

Primary Dns Suffix . . . . . . . :

Node Type . . . . . . . . . . . . : Unknown

IP Routing Enabled. . . . . . . . : No

WINS Proxy Enabled. . . . . . . . : No

DNS Suffix Search List. . . . . . : local.lan

Ethernet adapter Local Area Connection:

Connection-specific DNS Suffix . : local.lan

Description . . . . . . . . . . . : Intel(R) PRO/100 VE Network Connection

Physical Address. . . . . . . . . : 00-11-2F-28-29-E5

Dhcp Enabled. . . . . . . . . . . : Yes

Autoconfiguration Enabled . . . . : Yes

IP Address. . . . . . . . . . . . : 10.0.0.6

Subnet Mask . . . . . . . . . . . : 255.255.255.0

Default Gateway . . . . . . . . . : 10.0.0.2

DHCP Server . . . . . . . . . . . : 10.0.0.2

DNS Servers . . . . . . . . . . . : 10.0.0.2

Lease Obtained. . . . . . . . . . : Thursday, August 03, 2006 12:24:31 AM

Lease Expires . . . . . . . . . . : Thursday, August 03, 2006 12:24:31 PM

7.1.3 If your Internet/WAN access is the Static IP

Click on Start Menu -> Run .. -> enter command -> click OK -> ipconfig /all. The detail IP

configuration is showed as follow:

If the Dhcp Enabled is No in Ethernet adapter Local Area Connection, your WAN access is

the Static IP, please write down all the parameters (IP Address / Subnet Mask/ Default

Gateway / DNS Servers )for configuring the VoIP Gateway and then refer to the :

C:\Documents and Settings\ASUS P4P800VM>ipconfig /all

Windows IP Configuration

Host Name . . . . . . . . . . . . : asus

-00ef66a32b

Primary Dns Suffix . . . . . . . :

Node Type . . . . . . . . . . . . : Unknown

IP Routing Enabled. . . . . . . . : No

WINS Proxy Enabled. . . . . . . . : No

DNS Suffix Search List. . . . . . : local.lan

Ethernet adapter Local Area Connection:

Connection-specific DNS Suffix . : local.lan

Description . . . . . . . . . . . : Intel(R) PRO/100 VE Network Connection

Physical Address. . . . . . . . . : 00-11-2F-28-29-E5

Dhcp Enabled. . . . . . . . . . . : No

Autoconfiguration Enabled . . . . : Yes

IP Address. . . . . . . . . . . . : 10.0.0.6

Subnet Mask . . . . . . . . . . . : 255.255.255.0

Default Gateway . . . . . . . . . : 10.0.0.2

DNS Servers . . . . . . . . . . . : 192.168.10.100

Appendix A Glossary

This glossary defines acronyms and keywords used in this document.

A.1 Acronyms

ATA

Analog Telephony Adaptor

BLAM

Background Logging Application Mechanism

Broadband

Broad or wide bandwidth. In data transmssion, the wider the band, the

more data it is possible to transmit in a given time span. A cable, DSL and

ADSL connection to the network provide broadband for data transmission.

A dialup or ISDN connection typically provides a narrow bandwidth for

data transmission.

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" has hung up. It's

usually called "Open Loop Disconnect" when you're programming

telephone equipment. The CPC signal tells the phone equipment that the

outside party has hung-up, so it can stop recording to an answering

machine or voice mail, drop the call off hold, or just release a line that might

be used for dictation or announcements.

Generally speaking, if a human is using a phone line, it doesn't matter

whether the phone equipment recognizes CPC or not, since the human will

physically hang-up the phone when they're done with the call, or they'll pick

the call up off of hold when the phone system rings back after X seconds /

minutes.

CPC is normally sent as an open (0 volts DC), ranging from 250 to 500

milliseconds. When the outside party hangs-up, either on an inbound or

outbound call, the phone equipment sees this open on the line and hangs

up. Most voice mail and phone systems have a timer setting for CPC (or

Open Loop Disconnect). I generally set CPC at 500ms, unless I have a

problem. If you set it at 800ms, and the CPC open loop signal is only

500ms, the system will never see the open loop (it never gets to 800ms). If

you set it at 500ms, and the actual CPC duration is 800ms, the phone

system will recognize the CPC since there was 0 volts (an open loop) for

500ms (it won't matter if the open loop lasted another 300ms). If you

accidentally set it for 50ms you'll probably get cut-offs, especially during a

lightning storm which sometimes results in very brief blips in the loop

current. Setting this timer for 50ms means that if the phone equipment sees

an open for 1/20th of a second (not very long), it will hang up. Setting it for

500ms means it will hang-up if it sees an open of half a second or longer.

That's much more reliable.

There's often a short open (0 volts DC) on a phone line just after you go off-

hook, or just after you've finished dialing a phone number. These are

usually very short opens, like 20 to 50ms. If your phone system Open Loop

Disconnect timer is set at 50ms, you may never be able to make a call

because every call would be cut-off as soon as you went off-hook or were

finished dialing. That Open Loop Disconnect Timer is very

important!

DTMF

Dual-tone multifrequency. DTMF is the system that is used in interactive

voice-response menu systems such as the menu system for accessing

voicemail messages. The DTMF system allows the user to interact with the

menu by pressing keys on a dialpad or keyboard.

FoIP

Fax over Internet Protocol

FXO

Foreign Exchange Office

FXS

Foreign Exchange Station

IP

Internet Protocol. A data-oriented protocol used for communicating data

across a network. IP is the most common protocol used on the internet.

IP address

A unique number that devices use in order to identify and communicate

with each other on a computer network using the IP standard.

MWI

Message Waiting Indicator. An indicator that there is a voicemail message

for the owner of an account.

Narrowband

In data transmission, the wider the band, the more data it is possible to

transmit in a given time span. A cable, DSL and ADSL connection to the

network provide broadband for data transmission. A dialup or ISDN

connection typically provides a narrow bandwidth for data transmission.

PSTN

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

PTM

Packet Telephony Module RTP Real-time Transport Protocol

RFC

Request for Comment. A document that describes an aspect of an internet

technology.

An RFC may be a proposed, draft or full internet standard.

RTP

Real-time Transport Protocol. A protocol for delivering the media portion of

a data transmission over an IP network. SRTP is another media protocol.

Signaling

In a VoIP phone call, the information in a call that deals with establishing

and controling the connection, and managing the network. The non-

signaling portion of the call is the Media.

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 functions the user can access.

SIP address

The address used to connect to a SIP endpoint. In other words, the “phone

number” used in a VoIP phone call. For example, sip:[email protected].

STUN

Simple Transversal of UDP through NAT

TCP

Transmission Control Protocol. A transport protocol for delivering data

over an IP network. Other transport protocols are TLS and UDP.

TLS

Transport Layer Security. A transport protocol for delivering data over an

IP network. TLS is a secure transport protocol, which means that all the

data being transmitted (signaling and media) is encrypted. Other transport

protocols are TCP and UDP.

UA

User Agent

UDP

User Datagram Protocol. A transport protocol for delivering data over an IP

network. Other transport protocols are TCP and TLS.

URI

URI Uniform Resource Identifier. A name or address that identifies a

location on the world wide web. A SIP address is a type of URI.

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 for making phone calls over the

internet.

VoIP Service

Provider

A business that provides a VoIP service, allowing a user to connect to the

internet in order to make VoIP phone calls using VoIP Gateway. The VoIP

service provider sets up a SIP account for the user.

A.2 Keyword and Definitions

Caller

Call Originating End is called the Caller

Callee

The Call Terminating End is called the Callee

Transferor

The End transferring the call

Transferee

The End being transferred

Transfer Target

The End to whom the transferee is being

transferred

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