Well 3130IF User Manual

3130IF

---

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oIP

User

Manual

V3.0

1

3130IF

User

Manual

This

Manual

provides

basic

information

ononon

how

tototo

install

and

connect

3130IF

IPIPIP

Phone

tototo

the

network.

ItItIt

also

includes

features

and

functions

ofofof

3130IF

IPIPIP

phone

components,

and

how

tototo

use

them.

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111

Before

Getting

Started

Before you can connect 3130IF to the network and use it, you must have a high-

speed Internet connection installed. A high-speed connection includes such

environments as DSL, cable modem, and a leased line. 3130IF IP phone is a

stand-alone device, which re quires no PC to make Internet calls. 3130IF IP is fully

compatible with SIP and IAX2 industry standard and can interoperate with many

other SIP or IAX2 compliant devices and software in market.

222

Check

List

1) One 3130IF IP phone

2) One Straight Ethernet cable

3) Adapter

4) User Manual

333

Features

and

SPEC

3.1

Network

and

Protocol

IAX2

TCP(RFC793)

IP(RFC791)

UDP(RFC768)

ICMP(RFC792)

ARP(RFC826)

SNTP(RFC2030)

DHCP(RFC2131)

TFTP (RFC1350)

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DNS(RFC1034,RFC1035)

SIP RFC3261, RFC3262, RFC3666, RFC2543

-IP (RFC0791), TCP (RFC0793), UDP (RFC0768), ARP (RFC0826)

-ICMP, ARP, RARP, SNTP

-SDP TFTP DNS

-SIP server: 2 IP account with password protection

-Pubic Server/ Private server: can connect to ISP and Private SIP server at same

time

-Authentication: none, HTTP 1.1 basic/digest authentication for Web setup, MD5

(RFC2069/ RFC 2617)

-NAT traversal: STUN, NAT ALG, NAPT, SIP Express router

- 1 RJ-45 auto-sensing 10/100Mbps 。

-DHCP client (RFC2131)

/

TFTP client

/

PPPoE client

-Static IP

/

DHCP

/

PPPoE

-DNS client with 2 servers IP

3.2

Audio

Codec

G.711 µ

/

A

G.723.1

G.726

G.729

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3.3

QoS

- Q oS Diffserv, 802.1p/q

-VAD, CNG, Packet Loss Compensation, adaptive Jitter Buffer; Echo cancellation

G.168

3.4

Dial

Tone

Signal

Generation

DTMF (out of Band RFC 2833

/

in band)

SIP Info interoperate with CISCO SIP device

3.5

Phone

features

8 Kinds of ringer able select by number of Phone Box.

Flexible Dial Map: Fix length; End with #; Dial Map Table; Dial with time out

Redial/Hold

100 Phone book with Speed dial process

Do Not Disturb

Block call (by number)

Blind transfer

Forward (Off, Always, busy, No answer)

3.6

Configuration

Support

Web, Telnet, TFTP

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-Password protection for configuration

3.7

Firmware

Upgrade

-TFTP; FTP,

-Security firmware upgrade, firmware digest check

3.8

Log

-Log level

-Telnet logs and CDR

-SysLog Logs and CDR

-FTP/TFTP CDR upload

3.9

Physical

&&&

Environmental

Desktop

/

Wall mounting

Dimensions: 190X155X78 mm

Weight: 600 g (main unit)

Life-span:3 year

Operating Temperature: 0 ° to +40 ° C (32 ° to +104 ° F)

Storage Temperature:-20 ° to +70 ° C (-40 ° to +158 ° F)

Humidity: 5% -95% non-condensing

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444

Installation

Remove the LAN cable for Internet connection from your PC and connect it to

‘

WAN

’

port of 3130IF Connect the power adapter in the box to

‘

Power

’

Find LAN cable in the box and connect between

‘

Lan

’

port and your PC (PC is not

required for set up or making a call.)

555

Product

Overview

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666

Basic

Operations

6.1

Make

Phone

Calls

There are some ways to make phone calls:

1. Pick up handset or press H.F button, and then enter the phone numbers

2. Press the Redial button directly to redial the number last called.

Key

Button

Key

Button

Definitions

000

---

9,9,9,

*,*,*,

###

Digit, star and pound keys are also used for setting and call

process.

Local

IP/OK

D isplay local IP address on LCD

/

enter key

PPP

HHH

NNN

o./Edit

Browse phone number of unit

/

Configuration Modify

Vol

+++

Increase handset

/

speakerphone volume

Vol

---

Reduce handset

/

speakerphone volume

UpUpUp

Moving in menu

Down

Moving in menu

ESC/REG

Exit

/

register

Hold

Hold the active call/Delete

Speed

Dial

Dial stored speed dial number

Call

List

Browse call logs

Redial

Dial a new number or Redial the number last dialed.

H.F

Enter hands-free mode

Rec/Finish

Enter record menu

PHPHPH

Book

Enter phonebook operation

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3. While browsing in LOG menu, browsing OUTGOING

/

INCOMING call, press #

button to dial out displayed number on LCD.

4. When the unit indicates Missed calls, press 【 Call

Call

List

】

and 【 Down

Down

】

butto n to

enter Miss call menu, then press 【 Local

Local

IP/OK

】 button to review number. Pres s

【 # 】 button to dial out this number.

6. In ideal mode, enter the desired number to make pre-dial, Press 【 # 】 button

t

o

dial out this number, and press 【 Hold

Hold

】 delete and modify the number.

6.2

OGM

and

ICM

Unit can store 3 voice records which with max 80 seconds, so it can be configure

as 1 OGM and 2 ICM, while 2 ICM is full, unit will only play OGM while call

incoming till old ICM cleared.

To

play OGM and ICM recording, OGM shall be preset and “ Enable Voice Record

and Incoming Record Playing ” shall be set in menu. Then unit will play OGM after

5 ringings.

6.2.1

OGM

Record

and

playback:

Press 【 Rec/Finish

Rec/Finish

】 key and LCD show:

Press 【 Down

Down

】 or 【 UpUpUp

】 key and LCD show ”

---REC/FINISH record---

Received

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Press 【 Local

Local

IP/OK

】 key and 【 UpUpUp

】 key and LCD show:

Press 【 Local

Local

IP/OK

】 key and LCD show:

Press 【 Local

Local

IP/OK

】 key again to begin record and press “ Rec/Finish ” to quite

recording.

And press 【 DOWN

DOWN

】 or 【 UPUPUP

】 key and LCD show:

Press 【 Local

Local

IP/OK

】 key to playback User-Defined recording.

6.2.2

ICM

check

Press 【 Rec/Finish

Rec/Finish

】 key and LCD show:

Press 【 Local

Local

IP/OK

】 key and LCD show:

---REC/FINISH record---

User-Defined

--- User-Defined ---

Rec

Press OK to Rec

--- User-Defined ---

Play

---REC/FINISH record---

Received

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Press 【 Local

Local

IP/OK

】 key and LCD SHOW:

Press 【 Local

Local

IP/OK

】 key and LCD SHOW:

Press 【 Local

Local

IP/OK

】 key and LCD SHOW:

Press 【 Local

Local

IP/OK

】 key again to playback.

6.3

Phone

book

number

store

///

edit

///

delete

In standby, -press 【 Phone

Phone

Book

】 button, LCD display “ Phone Book Current ” -

press 【 OKOKOK

】 to check first record, press 【 UPUPUP

DOWN

】 to review others.

-press 【 UPUPUP

】 till LCD display ” ADD ”,then press 【 OKOKOK

】

,

input name, number,

position as LCD prompt. Press 【 HOLD

HOLD

】 to delete wrong digit.

- press 【 UPUPUP

】

till LCD display ” DEL ”,then press 【 OKOKOK

】

,

input position of desire

d

number and press 【 OKOKOK

】 to delete.

6.4

Speed

dial

number

from

Phone

Book

In standby, press 【 Speed

Speed

dial

dial】,

and input position number of phone book memo

ry,

press 【 ###

】 to dial out.

---Received---

New

--- New record ---

Record 1

--- Record 1---

Play

Press OK

To

Play

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6.5

call

list

check

///

delete

///

dial

out

In standby press button 【 Call

Call

list

】

,

LCD display ” Call Record Dialed ”

,

-Press 【 OKOKOK

】

to check last dialed out number and conversation time, press 【 UUU

PPP

】

and

【 DOWN

DOWN

】

to review other and press 【 ###

】

to dial out; press 【 HOLD

HOLD

】

to delete.

- press 【 UPUPUP

】 till LCD display ” Received ”,then press 【 OKOKOK

】 to check latest

received call, press 【 UPUPUP

】

and 【 DOWN

DOWN

】

to review other and press ” # ” to dial out;

press 【 HOLD

HOLD

】 to delete.

- press 【 UPUPUP

】 till LCD display ” ” Call Record Dialed ”,then press 【 OKOKOK

】 to check

latest miss call, press 【 UPUPUP

】

and

【 DOW

DOW

N

】

to review other and press ” # ” to dial out;

press 【 HOLD

HOLD

】 to delete.

6.6

hold

During conversation, press 【 Hold

Hold

】 to keep line and press again to release.

6.7

Volume

adjustment

ininin

conversation

During conversation, press 【

Vol

+/-

】 to adjust receiving volume.

6.8

Block

list

setting

///

edit

///

delete

Please refer to below setting menu and CALL SERVICE setting in web

configuration.

6.9

3-party

conference

During conversation, press 【 HOLD

HOLD

】

,

then dial another number plus # button, wh ile

line connected, press 【 HOLD

HOLD

】 and 【 * 】 to make conference call

6.10

call

forwarding

：：：

During conversation, press 【 SPEED

SPEED

DIAL

】

,

then dial another number plus #

button to forward it.

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777

Unit

Configuration

777

.1.1.1

IPIPIP

distribution

mode

selection:

Press and hold 【 111

】 button for 5s, the LCD displays “

STATIC

MODE ”

;

Press and hold 【 2 】 button for 5s, the LCD display “ DHCP MODE ”

;

Press and hold 【 3 】 button for 5s, the LCD display “ PPPOE MODE ”

.

777

.2.2.2

Configuration

with

keypad

and

LCD

display

In standby, press and hold 【 # 】 button till LCD shows " Input Password: " input

correct password (default is 123), press 【 OKOKOK

】 key to enter the menu list. Then

follow below menu list to set parameters accordingly.

During configuration, operations as follows

-For browse and edit Configuration, press 【 UPUPUP

】

,

【 DOWN

DOWN

】

;

-To change parameter, press 【 Phone

Phone

NoNoNo

】

firstly, then input desired digit, confir m

and save by press button 【 Local/OK

Local/OK

】

.

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For

example:

Account below

login: xxxxxxxx

pw: yyyyyy

Sip server :aaa.bbb.ccc

Not stun

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Input Password

:

Congfig

Network

Congfig

SIP

Reg Status

Press and hold

“ # ”

key 3Sec

Input “123

”

Press “OK

”

Keep press

“

UP

”

Press “OK

”

Press “UP

”

SIP

Stun

Press “OK

”

Stun

Switch

Press “OK

”

Switch 0 [

OFF

]

Press “Phone No

”

Please select :

Input 1 or

0

Press “OK

”

Press “UP

”

Press “ESC

”

SIP

Signal-port

Press “OK

”

Signal-port 5060

Press “Phone No

”

Input Value

:

Input SIP port

Press “OK

”

Press “ESC

”

Press “UP

”

SIP

RFC-version

Press “UP

”

SIP

Swap-server

Press “OK

”

Swap-server 0 [

OFF

]

Press “Phone No

”

Please select :

Input 1 or

0

Press “OK

”

Press “ESC

”

Press “UP

”

SIP

Interval-time

Press “UP

”

SIP

Dtmf-mode

Press “OK

”

Dtmf-mode 0 [

DTMF_RELAY

]

Press “Phone No

”

Please select :

Input Dtmf-mode

Press “OK

”

Press “ESC

”

Press “UP

”

SIP

Detect-server

Press “OK

”

Detect-server 0 [

OFF

]

Press “Phone No

”

Please select :

Input

1

Press “OK

”

Press “ESC

”

Press “UP

”

SIP

User Agent

Press “UP

”

SIP

Domain

Press “OK

”

Domain

Public

Press “OK

”

Public

Press “Phone No

”

Input Domain

Press “OK

”

Input Value

Press “ESC”2

times

Press “UP

”

SIP

Server

Press “OK

”

4

times

Addr

Press “Phone No

”

Input Value

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Register

Expire Time

Register

Password

Input SIP Server Addr

“

Aaa.bbb.ccc

”

K press “UP

”

Press “OK

”

Input Value

:

Press “OK

”

Press “OK

”

Phone-number

Public SIP

Press “OK

”

Press “OK

”

Press “ESC

”

Press “Phone No

”

Input Password

“

yyyyyy

”

Press “ESC

”

Press “UP

”

Register

User

Press “OK

”

Input Value

:

Input SIP User

“

xxxxxxxx

”

Press “ESC

”

4

times

Press “UP

”

SIP

Reg Switch

Register 0 [

OFF

]

Press “Phone No

”

Please select :

Input

“ 1 ”

Press “OK

”

Press “ESC

”

3

times

Press “OK

”

2

times

Config

SIP

Keep press “UP

”

Public SIP

Press “Phone No

”

Input number :

Input SIP User

“

xxxxxxxx

”

Press “OK

”

Press “ESC

”

5

times

Press “Down”2

times

User

Press “Phone No

”

Config

Call Feature

Press “UP

”

2

times

Phone-number

SIP

Press “OK

”

2

times

Config

System

Press “OK

”

Press “OK

”

System

Save

Press “OK

”

Are you sure

?

Press “OK

”

Press “ESC

”

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Note:

1.1.1.

Stun

Config

default

value

isisis

000

[OFF]

,,,

don

’’’

ttt

modify

normal.

2.2.2.

Signal-port

default

value

isisis

5060,

don

’’’

ttt

modify

normal.

3.3.3.

Interval-time

default

value

isisis

606060

sec,

don

’’’

ttt

modify

normal.

888

Configuration

via

WEB

The IP Phone Web Configuration Menu can be accessed by the following URI:

Pick up handset and input “ # * 111# “,you will hear the “ IP address ”

888

.1.1.1

logon

Web

While input correct IP address as above, logon menu pop out as follows:

There are two level login as:

-Guest account: the default username and password is “ guest ”

,

user can have a

browse of system.

-Administrator account: the default username and password is “ admin ”

,

this user

can configure the system.

Note:

After

inputting

username

and

password,

user

press

carriage

return

directly

tototo

enter

the

page.

While successfully login, web shown as follows:

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888

.2.2.2

Current

state

On this page user can gather information of each normal parameters, as:

-the network section shows the current WAN of the phone, including access way of

WAN IP and

IP

（ static state,

DHCP,

PPPoE

）

,

MAC address, WAN IP address ofthe

phone

.

-

The VoIP section shows the current default signaling protocol, and server

parameter, whether enables register, whether has registered on GK; Register

server IP of

SIP,

proxy server

IP,

whether enables register, whether has registered

on register server, w hether enables outbound proxy, whether enables STUN server .

-The Phone Number section shows corresponding phone number of each protocol;

the version number and date of issue have been shown at the end of the page.

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888

.3.3.3

Network

configuration

8.3.1

Wide

area

network

（（（

WAN

）））

User can view the current network IP linking mode of the system on this page.

User will be authorized to set the network

IP,

Gateway and DNS if the system

adopts the static linking mode.

If the system selects DHCP service in the network which is using DHCP service, IP

address will be gained dynamically.

If the system selects PPPOE service in the network which is using the PPPOE

service, then the IP address will be gained by the set PPPOE ISP internet and

password of the account.

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Note:

IfIfIf

IPIPIP

address

has

been

modified,

the

web

page

will

nonono

longer

respond

owing

tototo

the

modification,

sososo

new

IPIPIP

address

should

bebebe

input

ininin

the

address

field

now.

Configuration Explanation:

Current phone

IP,

subnet mask, mac address and current phone IP;

,

Select acquisition way of IP for WAN; This is single option; Configure static IP

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parameter for WAN;

Configure static IP address;

Configure subnet mask;

Configure IP address of the phone;

Configure “ DNS domain ” suffix; if user input “ domain ” and it can

’ t

be resolved, then

the phone will add and resolve the “ domain ” after user has input;

Main DNS server IP address;

The second DNS server IP address;

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Configure PPPoE:

Service name, if PPPoE ISP has no special requirement for this name, generally is

the default;

PPPoE account;

PPPoE password;

Configure the parameter and then click “ apply ” to go into effect.

8.3.2SNTP

configuration.

On this page, user can save and configure time zone setting.

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Configure the desired time zone.

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8.4

VOIP

configuration

888

.4.1

SIP

configuration.

Configuration Explanation:

show SIP register state; if register successfully, there will show Registered in the

square bracket, otherwise show Unregistered;

Configure SIP register server IP address;

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Configure SIP register server signal port;

Configure SIP register account （ usually it is the same with the port number that

configured, some special SIP servers will have different port configurations, then

the port configuration needs to be configured to be numbers, here the configuration

account can be arbitrary character string ）

;

Configure password of SIP register account;

Configure proxy server IP address （ usually SIP will provide user with service of

proxy server and register server which have the same configuration, so the

configuration of proxy server is usually the same with that of register server, but if

the configura tions of them are different(such as different IP addresses), then each

server's configuration should be modified separately ）

;

Configure SIP proxy server signal port;

Configure proxy server account;

Configure proxy server password;

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Configure local signal port, the default is 5060 （ this port will go into effect

immediately, the SIP call will use the modified port for communication after

modification ）

;

Configure expire time of SIP server register, the default is 600 seconds. If the

expire time that server requires is more or less than that configured by the phone,

the phone can automatically modify it to the recommended time limit and register;

Configure detection interval time of the server, if the phone enables SIP detection

server function, the phone will detect once for whether the server has response

every other detection interval time;

DTMF sending mode configuration; three kinds: the above are basic configurations

of

SIP.

Configure SIP of the phone as default protocol;

Enable the phone to use protocol edition. When the phone need to communicate

with phones which is using SIP1.0 such as CISCO5300 and so on, then it should

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be configured into RFC2543 to communicate normally. the default is to enable

RFC3261;

Configure enable/disable register;

SOME ISP INTERNET MAY INHIBIT THE PHONE TO REGISTER AND CANCEL

THE REGISTER IN SUCCESSION, SO USER HAD BETTER NOT APPLY OR

REGISTER AND CANCEL SOON IN SUCCESSION AND SUBMIT

REGISTRATION

REPEATEDLY.

SERVER

MAY

STOP RESPONSE OF

DIALOGUE MACHINE, THEN THE PH ONE RECEIVES NO CERTIFICATION OF

REGISTER/CANCEL LOGIN REQUEST AND REGISTRATION

STATE

WILL

SHOW AS INCORRECT!

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888

.4.2

IAX2

Configuration

Configuration Explanation:

IAX2 registration

state display ； If register successfully, it will display [Registered], otherwise will

display [Unregistered];

Configuration iax2 the server address, also can use domain name form;

Configuration iax2 server port;

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Configuration IAX2 account name ；

Configuration IAX2 account password ；

Configuration IAX2 phone number ；

Configuration equipment iax2 monitor port ；

Configuration voice mail number，If the iax support voice mailbox, the voice mailbo x

is the letter form, the gateway is unable to input the letter, uses this number to

replace voice mail the name ；

Configuration voice mailbox name; if the iax support voice mailbox, here to

configuration the voice mailbox the name ；

Configuration whether supports echo. If the platform support echo, (echo number is

the text format), then the telephone configuration this echo test number replace

echo actual text number. This function is refers through the platform, the terminal

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may carry on echo the call to test.

To

see the terminal to the platform converses on

the telephone whether normally;

Configurationecho test text ；

Configuration iax refresh time ， The unit of time for the second, suggested the user

in makes the choice 60 to 3600 between ；

configuration the permission/prohibition registers the server ；

Configuration whether supports G.729, if the platform supports 729, the telephone

transmits codec supports 729, then calls idefisk to be able to cause idefisk as a

result of to cause pc to 729 support questions to shut down;

Configuration the iax2 protocol for the default call protocol. After chooses this, the

user picks the call which machine sends out to be able through the iax2 protocol to

carry on the correspondence. The gateway default is the SIP protocol, this

configura tion mainly aims at the host to call, is not called this influence. If after the

user configuration this item, but picked machine can use iax no longer is

SIP,

if the

user also wanted simultaneously SIP to send out the call, then might dispose dial

peer the prefix replace function to realize the SIP call;

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8.4.3

STUN

Configuration

Configure explanation of private server;

To

show the phone whether has been registered on public server

or

private server;

Configure IP address of SIP STUN server;

Configure port of SIP STUN.

STUN can support SIP terminal's penetration to

NAT

in the inner-net. In this way,

as long as there is conventional SIP proxy and a STUN server placed in the public

net, it will do; but STUN only supports three

NAT

modes: FULL CONE, restricted,

port restri cted;

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Private server configuration. specific configuration parameter has the same

meaning with public server;

Interval time for STUN's detection on

NAT

type, the unit is minute;

Configure enable/disable PRACK

Configure automatic detection server of the phone;

Configure enable/disable PRACK

Configure enable/disable SIP STUN;

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Configure permit/deny private server register;

Note:

IfIfIf

you

want

tototo

register

and

call

through

server,

you

must

configure

corresponding

numbers

（ which

which

are

usually

SIP

accounts

） tototo

local

port,

Otherwise

the

phone

will

reject

for

sending

out

register

message

when

ititit

considers

that

there

isisis

nonono

number.

After the aforesaid network and VoIP configurations have been configurated on the

phone and internetwork communication has been implemented, the user can make

VoIP calls by the calling register and proxy.

8.4.4

Number

binding

configuration

Function of number IP table is one way to implement the phone's calling online, and

the calling of the phone will be more flexible by configuration the number IP table.

For example, user know the other party's number and IP and want to make direct

call

t o

the party by point-to-point mode: the other party's number is 1234,make a

configuration of 1234 directly ,then the phone will send the called number1234 to

the corresponding IP address; Or set numbers with prefix matching pattern, for

example, user want to make a call to a number in a certain region （ 010 ） ,user can

configure the corresponding number IP as 010T ― protocol ―

IP,

after that,

whenever user dial numbers with 010 prefix( such as 010 － 62201234),the call will

be made by this rule.

Bases on this configuration, we can also make the phone use different accounts

and run speed calling without swap.

When making deletion

or

modification, select the number first and click load, then

click Modify and complete the operation.

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8.4.5

DIAL

PEER

Configuration

Explanation:

It is to add outgoing call number, there are two kinds of outgoing call number setup:

One is exactitude matching, after this configuration has been done, when the

number is totally the same with the user's calling number, the phone will make the

call with this number's IP address image

or

configuration; Another is prefix

matching （ be equivalent to PSTN's district number prefix function

）

,if the previou

s

A

bits of this number are the same with that of the user's calling number(the prefix

number length),then the phone will use this number's IP address image

or

configuration to make the call. When configuring the prefix matching, letter "T"

should be added b ehind the prefix number to be distinguished from the exactitude

matching; the longest length is 30 bits.

Configure the calling mode: SIP and IAX2

;

Configure destination address, if it is point-to-point call, then input the opposite

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terminal's IP address, it can also be set as domain name and resolved the specific

IP address by DNS server of the phone. If no configuration has been made, then

the IP will be considered as 0.0.0.0. This is an optional configuration item;

Configure the other party's protocol signal port, this is optional configuration item

:when nothing is input, then the default of h323 protocol is 1720,the default of sip

protocol is 5060;lifeline required no configuration of this item, shown as 0;

Configure alias, this is optional configuration item: it is the number to be used when

the other party's number has prefix; when no configuration has been made, shown

as no alias;

Configure suffix ,this is optional configuration item: it is the additive dial-out number

behind the number; when no configuration has been made, shown as no suffix;

Configure the replacing length, replace the number that user input according to this

length; this is optional configuration item.

Of which the alias can be divided into four types, it should be combined with

replacing length to make the setup:

Add: xxx, add xxx before number. in this way it can help user save the dialing

length;

All: xxx, the number is all replaced by xxx; speed dialing can be implemented, for

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example, user configure the dialing number as with the configuration "all"

,

the

actual calling number will be replaced;

Del: delete

n

bit in the front part of the number,

n

can be decided by the replacing

length; this configuration can decide the protocol for appointed number;

Rep: xxx,

n

bit in the front part of the number will be replaced.

n

is decided by the

replacing length. For example, user want to dial

PSTN

（ 010 － 62281493

）

by VoIP'

s

Rec/Finish over service, while actually the called number should be 8610 －

62281493,then we ca

n

configure called number as 010T ,then rep:8610,and then

set the replacing length as 3. So that when user make a call with 010 prefix, the

number will be replaced as 8610 plus the number and then sent out. It is a

convenient thinking mode for user to mak

e

a call;

Delete selective number IP image;

If user want to modify a certain current number image, first select

in the drop-down menu and then load the image parameter of the said number, click

modify to make modification; of which:

this is the modified number. read-only;

To

modify call mode;

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To

modify destination address; this is optional configuration item;

To

modify destination phone port;this is optional configuration item;

To

modify alias; this is optional configuration item;

To

modify suffix; this is optional configuration item;

To

modify replacing length （ if rep and del of alias have been configured ）

;

Click submit to go into effect; The basic application of the number IP table has been

introduced, now let me introduce how to configure IP table of number to implement

configuration of using multi-accounts concurrently:

For example, now user has a H323 account and two SIP accounts, then under the

default condition, user can only make calls by the default protocol. Configure the

number IP table to select the call protocol, then user don

’ t

need to select default

protocol b efore making calls every time.

The configuration process will not be repeated, now

I

will mainly introduce what

kind of number IP image can implement this function.

By configuration, Image table as follows will be gained:

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Image of 9T means when user configure public SIP server and register, then user

just need to add a"9"before the calling number whenever making a call by public

SIP;

Image of 8T means when user configure private server and register, then user just

need to add a"8"before the calling number whenever making a call by private SIP;

Image of 7T means when user configure h323 server and register, then user just

need to add a"7"before the calling number whenever making a call by H323 GK.

888

.5.5.5

Phone

configuration

8.5.1

DSP

configuration

On this page, user can set speech coding, IO volume control, cue tone standard,

caller

ID

standard and so on.

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Configuration

Explanation:

Configure ring type

Configure output volume;

Configure input volume;

Configure handsfree volume;

Configure handdown time, that is, if the hooking time is shorter than this time, then

the gateway will not consider the user has handdown.

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8.5.2

Call

service

configuration

On this page, user can set value added services such as hot-line, call forwarding,

call transfer (CT), call-waiting service three way call, blacklist, out-limit list and so

on.

Configuration

Explanation:

Configure hot-line number of the port. With this number of the port, this hot-line

number will be dialed automatically as soon as off-hook and user can's dial any

other number;

number IP configuration of call transfer (CT);

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Configure enable/disable call waiting service; after it is enabled, user can hold

calls of the other party by hooking, with hooking again, the hold call can go on;

Configure enable/disable call transfer (CT); after it is enabled, there are two modes

call transfer as below:

UNATTENDED TRANSFER: During conversation, press 【 SPEED

SPEED

DIAL

】 button

and input transferred number end with 【 ###

】 to transfer the phone to the third part

and hang up automatically

ATTENDED TRANSFER: During conversation, press 【 HOLD

HOLD

】

button to hold thi

s

line, and input transferred number end with 【 # 】 to get through another line. After

conversation with third part, press 【 SPEED

SPEED

DIAL

】

button to end conversation a nd

transfer the phone to the third part and hang up automatically.

Configure enable/disable three way call; user can call the other part as the call

origination, after talking, make hooking to hold this part and then press 【 *

】

key

t o

hear the dialing tone, after call completion to the third party, hooking again to

recover the talk with the second part, then the three way call concurrently;

Configure enable/disable Enable Rec/Finish Record, then no body answer the call,

the phone will into the answering function;

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Configure enable/disable User-Defined Rec/Finish, then enable Rec/Finish record,

the phone will auto request user leave message.

After the aforesaid configuration has been done, click apply to make them go into

effect.

Configure add/delete blacklist. If user doesn't want to answer a certain number,

please add this number to the list, and then this number will be unable to get

through the phone.

Configure out-limit list; for example, if user don't want the phone to dial a certain

number, please add the number to this table, and the user will be unable to get

through this number.

8.5.3

Phone

book

configuration

On this page, user can save and configure telephone book.

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Additional Phone book configure

Configure the Name

/

Number

/

Ring type of Phone Book.

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888

.6.6.6

MAINTENANCE

configuration

8.6.1

Auto

PROVISION

Unit can be set as automatically upgrade from desired FTP

or

TFTP server.

Configure unit as follow steps:

Input IP add. of desired FTP server.

Input user name of desired FTP server.

Input user password of desired FTP server.

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Input name configuration file. Software version must be different for each upgrade

file.

If configuration file is encrypted, password needed.

Chose server type as either FTP

or TFTP.

Set auto-upgrade interval duration.

Chose auto-upgrade type.

8.6.2

Save

and

Clear

configuration

User can save the current configuration on this page.

The system configuration can be set as factory default configuration on clear config

page and the phone will restart automatically.

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888

...

666

...

333

Upgrade

on-line

Upload

WEB

page

OnOnOn

this

page,

user

can

select

the

upgrade

document

(firmware

ororor

config

file)

ononon

hard

disk

ofofof

the

computer

directly

tototo

run

the

system

upgrade.

After

the

upgrade

has

been

completed,

restart

the

phone

and

ititit

will

bebebe

usable

atatat

once.

FTP

download

On this page, user can upgrade system and configure files by FTP

or

TFTP mode.

Configuration

Explanation:

Configure upload

or

download FTP/ TFTP server IP address;

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Configure username of the upload

or

download FTP server. If user select TFTP

mode, username and password are not required to be configured;

Configure upload

or

download of FTP server password;

Configure upload

or

download system upgrade document

or

system layout file

name. It should be noted that system file take .dlf as suffix, configuration files take

.cfg as suffix;

Select server type;

Click image update button, the phone will upgrade system file;

Click config upload button, the phone will upload its configuration files to FTP/TFTP

server and save with names of user-defined configuration files;

Click config

config

download

button, the phone will download configuration files of

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FTP/TFTP server to the phone and the configuration will go into effect after

restarting;

Output configure file can be edit, delete,

or

make comment starting by # on

each command. Unit support module upgrade, like if changes made to SIP

configure, others in configure file can be deleted and configuration in unit will not be

affected.

While upgrade unit with modified configure file, please make sure check each

parameter while finished upgrade. In case of anything wrong, please recover

configure under POST mode.

Configure

file

encryption

Configure file can be encryption with DOS command:

dsc.exe <key.txt> <e/d> <old configure> <new configure>.

Dsc.exe-encryption software tool

<key.txt>-user made encryption key file

<e/d> e （ encrypt ）

, d

(decrypt)

< old configure >former configure file name and path,

< new configure >new configure file name, defined by user.

Configuration

files

WEB

download

On this page, user can directly select the configuration files on the hard disk of the

computer, and then make modification to the system configuration, after the

download, restart the phone and the configuration will go into effect.

8.6.4

Account

management

On this page, user can add and delete users according to own needs and can

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modify user's authorities there have been.

Configuration

Explanation:

display of phone user account list;

To

Add Phone Account:

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account level; root possesses

authorities to modify configuration,

general possesses read-only authority;

as corresponding password of the additive

account; As second confirmation of password, to ensure correct

setup of password;

Click submit to go into effect; click return to

cancel configuration and return.

Select users that you want to delete in the drop-down menu, click Delete.

To

modify the chosen accounts, need to select account first, click load again and

then click modify, it will be shown at lower part of page as the following figure, of

which:

The modified username;

Modify user authorities;

Modify user password;

Make confirmation of the

modified user password;

Submit

or

cancel the modification;

Owing to the phone's default account: accounts of the administrator level-admin

and the ordinary level － guest are

all

weak account and weak password, the

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username and password will be easily to be guessed on public network, so the user

had better modify the administrator and ordinary user.

Enter with manager level when making modification, create a administrator account

and a browse account (you'd better not set the name as admin, administrator, guest ,

etc.),set password and then save configuration, entering with new manager

account, delete default manager and browse account and save configuration,

security will be enhanced!

888

...

777

System

management

VPN

configuration

On this page, user can save and configure VPN setting.

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Configure VPN server address;

Configure VPN server port;

Configure VPN server group ID;

Configure VPN server area code;

Configure enable/disable VPN tunnel;

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8.8

Logout

configuration