AT-640 User Manual

AT-640 User Manual

ISSUE 1.0

2011-07-28

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AT-640 User Manual

1st、 AT-640’s Network Features...........................................................................4

1、 The View.........................................................................................................................4
2、 Interfaces.......................................................................................................................5
3、 Hardware....................................................................................................................... 5
4、 Software......................................................................................................................... 5
5、 Network.......................................................................................................................... 6
6、 Management and Maintenance..............................................................................6
7、 Protocol...........................................................................................................................6
8、 Compliant Standard................................................................................................... 7
9、 Operating Requirement............................................................................................ 7
10、 Packing List................................................................................................................... 7
11、 Installation.................................................................................................................... 7

2nd、 Feet installation instruction................................................................... 8

3rd、 Keypad of IP Phone.................................................................................11

4th、 BLF indicator status................................................................................ 12

5th、 Menu Operation........................................................................................13

6th、 Basic functions and operations.......................................................... 18

1、 Answer the calls........................................................................................................ 18
2、 Make Call......................................................................................................................19
3、 Speed dial....................................................................................................................19
4、 Multiple line dial-up................................................................................................. 20
5、 Hang up the phone.................................................................................................. 20
6、 Call Transfer................................................................................................................20
7、 Call Hold....................................................................................................................... 21
8、 3-Way Conference Calls......................................................................................... 22
9、 Call History..................................................................................................................22
10、 Call pickup................................................................................................................... 23
11、 Join call.........................................................................................................................23
12、 Redial/Unredial.......................................................................................................... 23
13、 vport.............................................................................................................................. 24
14、 Click to dial..................................................................................................................25
15、 SMS function.............................................................................................................. 25
16、 Default Password...................................................................................................... 26
17、 Check the Phone’s IP...............................................................................................26

7th、 Web settings.............................................................................................. 26

1、 Current state.............................................................................................................. 27
2、 Network........................................................................................................................ 27

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2.1. Wan Config..................................................................................................27

2.2. LAN Config.................................................................................................. 29
3、 VoIP................................................................................................................................30
3.1. SIP1/SIP2/SIP3/SIP4..............................................................................30
3.2. Iax2 Config................................................................................................. 33
4、 Advanced..................................................................................................................... 34
4.1. DHCP Server.............................................................................................. 34
4.2. NAT................................................................................................................ 35
4.3. Net Service................................................................................................. 38
4.4. Firewall......................................................................................................... 38
4.5. Qos.................................................................................................................39
4.6. Digital Map...................................................................................................41
4.7. Stun............................................................................................................... 42
4.8. Call Service.................................................................................................43
4.9. MMI Filter.................................................................................................... 44
4.10. Audio Settings........................................................................................... 45
4.11. VPN................................................................................................................ 46
5、 Dial Peer....................................................................................................................... 47
6、 Config Manage........................................................................................................... 49
7、 Update...........................................................................................................................49
7.1. Web Update................................................................................................ 49
7.2. FTP/TFTP Update...................................................................................... 50
7.3. Auto Provisioning...................................................................................... 51
8、 System Manage......................................................................................................... 52
8.1. Account Manage........................................................................................52
8.2. Phone Book.................................................................................................54
8.3. Multi Line Set............................................................................................. 54
8.4. Function Key Set.......................................................................................55
8.5. Syslog Config............................................................................................. 57
8.6. Time Set...................................................................................................... 58
8.7. Call Log........................................................................................................ 59
8.8. Language Set............................................................................................. 59
8.9. Logout...........................................................................................................60
8.10. Reboot.......................................................................................................... 60

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1st
1st、 AT-640’’s Network Features
AT-640

1、The View

AT640

AT640 with expander board

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2、Interfaces

� Power: Output Power: 12VDC, 500mA DC

� WAN: RJ45 port
� LAN: RJ45 port
� Extended interface for BLF module: 2
� Headset jack : RJ9 port
� Handset jack : RJ9 port

3、Hardware

� LCD: 128×64 dot matrix

� FLASH: 4M
� RAM: 16×16M
� LED indicator ： 1 Status Light , 9 BLF indicator, 1 voicemail indicator, 1
headset indicator, 1 mute indicator, 1 handfree indicator

4、Software

� Sip 2.0 (RFC3261) and other related SIP RFC

� 4 lines SIP
� 1 line IAX2
� STUN
� Jitter Buffer(200ms),VAD,CNG
� G.711A/u, G722, G.723, G.726-32, G.729 Codec
� G.168 compliant 96ms echo cancellation
� Support SIP domain，SIP authentication(none，basic,MD5),NDS.
� Support inbound audio, RFC2833 and SIP info , DTMF transmission way
� SIP Call Forward、Call transfer、Call hold、Call waiting, 3-way talking、
Pickup、Join call、Redial、Unredial、Call Park、Vport、Click to dial
� Dial without register
� Support Hotline、DND(Do Not Disturb)、Blacklists、Call Limitation、Caller ID
� Dial-peer calling rule, IP to IP call
� SIP server conference
� Phone book with 500 records, 100 answered call、100 missed call
� Support HTTP、FTP TFTP updating the configuration and firmware
� Syslog
� Auto answer
� Support SNTP client
� Telnet, WEB visit terminal
� Support different level user management

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� Support multi language ( LCD support Latin language system, web support
all languages)
� soft button: soft button * 4
� Support BLF/BLA
� Support SMS
� Support auto provision (option 66) for config file upgrade

5、Network

� WAN/LAN：Support bridge and route mode

� Support base of NAT and NAPT
� Support PPPoE，（ADSL，cable modem use for internet connecting）
� Support VLAN（DATA VLAN and VOICE VLAN）
� Support DMZ
� Support L2TP VPN（OpenVPN optional）
� WAN support Primary and Alter function
� WAN support DHCP Client
� LAN support DHCP Server
� Qos support Diffserv
� Support Network command tool: include ping, trace route, telnet

6、Management and Maintenance

� Support safe mode(POST Mode) and firmware updating under safe mode
� Support different level user management
� Configuration via web , keyboard and command
� Support multi language (LCD support Latin language system, web support
all languages) and easy dynamic switch between different languages
� Firmware and configuration updating via HTTP , FTP and TFTP
� Support system log and call log
� Firmware and configuration file auto provision

7、Protocol

� IEEE 802.3 /802.3 u 10 Base T / 100Base TX

� PPPoE：PPP over Ethernet
� WAN support DHCP Client and Server
� SIP RFC3261, RFC3262, RFC3263, RFC3264, RFC3265, RFC2543, RFC3489,
RFC3842, RFC3515, RFC2976, RFC3428, RFC2327, RFC2782, RFC1889
� TCP/IP: Transfer Control Protocol/Internet Protocol
� RTP: Real-time Transport Protocol

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� RTCP：RTP Control Protocol

� VAD/CNG
� Telnet：remote host access protocol
� DNS: Domain Name Server
� TFTP：Trivial File Transfer Protocol
� HTTP：Hypertext Transfer Protocol
� FTP：File Transfer Protocol

8、Compliant Standard

� CE：EN55024,EN55022
� FCC part15
� Comply with ROHS in EU
� Comply with ROHS in China

�
Explanation:
The letter “e” is the first letter of “environment: and “electronic”. The rim is a
round with two arrow, stands for recycle. The number 20 stands for the years of
environment protection. Please note the years of environment protection is not
discarding year nor usage life.

9、Operating Requirement

� Operation temperature: 0 to 40° C (32° to 104° F)

� Storage temperature: -30° to 65° C (-22° to 149° F)
� Humidity: 10 to 90% no dew

10
10、 Packing List

� AT-640 IP phone
� Power adaptor (output 12v ,500mA)
� Manual CD

11
11、 Installation

Use Ethernet cable to connect AT-640’s LAN port and your computer. Set computer’s
IP to the network 192.168.10.x or using dynamic obtain IP. Open web browser and
key in 192.168.10.1. Then user will see the login page of AT-640, the default

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username and password is admin/admin for administrator and guest/guest for

guest.
Set up page for VoIP user only:

AT640 WAN default to use DHCP, and after connect WAN to Switch, it will obtain IP
automatically. Press the down key of navigation button, you will see AT640's IP.
Then type the ip in your web browser, you will see the login page.

2nd
2nd、 Feet installation instruction

1、Desktop position:
A、Put the bottom side of the IP phone upside and press the plate with letter
“PUSH” into the slot, please refer the picture as below:

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B、Press the other plate into the slot in accordance with the direction of the
arrow

C、Repeat A and B. It is the right picture of putting on desk after fixing the two
feet below:

D、 Disassemble the feet:

Press the plate with word “PUSH” and pull the feet with the direction of arrow.
When the plate is pull out of the slot (there will be a sound of “pa”) you can
take off the feet

2、On wall postion

A、Put the bottom side of the IP phone upside and push the plate with letter
“PUSH” into the slot, please refer the picture as below:

B、Push the other plate into the slot in accordance with the direction of the arrow

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C、Repeat A and B. It is the picture of wall mounting after fixing the two feet
below:

Attention: Please rotate the hook to the position as in

picture with a coin or other tools

D、 Disassemble the feet way:

Press the plate with word “PUSH” and pull the feet with the direction of arrow.
When the plate is pull out of the slot (there will be a sound of “pa”) you can
take off the feet

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3rd
3rd、 Keypad of IP Phone

Describe of the buttons and Screen:

Soft buttons Press to select an feature shown in the soft button features
Soft button Shows available choices based on current phone function
features displayed on the last line of LCD screen
Status Shows the phone status, if the phone is standby, the LED is
with light. If there is income calling, the LED will flicker.
� If the phone is starting ,the LED is flicker
� If the phone is standby, the LED is off
� If there is income calling, the LED will flicker. The
frequency is 500ms off, 500ms on.
� When have voicemail, LED shows red and flicker, and the
frequency is 1000ms off, 1000ms on.
� If the phone not obtain the IP address, the LED is ON

LCD Screen Display screen for the phone: It shows the date, time, phone
number, incoming caller’s ID(if available),line/call status,

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extension numbers and the soft button features.

Lines Shows extension number and status.
There are three colors for LED, red，yellow and orange.
� If the line is registered, the LED shows yellow
� If the line is enable registered but register to server failed, the
LED shows orange
� If the line has income calling, the LED shows red and flicker
� If the line is on the calling , the LED shows red
� If the line disable for register, the LED is off.
� when there is the incoming call , LED blinks, The frequency is
500ms off,500ms on.
� When have voicemail, LED shows red and flicker , The frequency is
1000ms off,1000ms on.

Navigation Allows users to navigate(left, right, up, down), on the

button standby, up and down shows the network information, right
shows the lines information, left shows the call record
Dial pad For entering numbers, letters or characters
Function Key All the function key can be defined as memory key/line/key
event to realize different function.
Menu Come into Keypad menu
Volume buttons Adjust the volume
Speakerphone Pick up and hung up on the speakerphone mode, when pick
button up by speakerphone, the LED of the button is on
Mute button Mute the handset, headset or speakerphone by press the
Mute button; this prevents the person on the active call
form hearing what you or someone else in the room is
saying.
To cancel the Mute function, press the Mute button again.
If Mute the voice, the LED is light on this button
Headset button Pick up and hung up on headset mode. When pick up by
headset, the LED button will light
Voicemail button Check the Voicemail status, if there are voicemail, the button
will light

4th
4th、 BLF indicator status

1.AT640 support BLF in two ways, one is integrated on the ipphone, the other one is
extended interface.
The BLF indicator status:

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� When boot up,all the light blink for once.

� When phone is stand by,all the light are off.
� When there is a coming call,the corresponding line key and status
light will blink.

5th
5th、 Menu Operation

Head Menu Secondary menu Level 3 Menu Level 4 Menu

When Stand by
Dial
Next
Dial
OK
Quit
DIR
(phone book)
List Is Empty
(when no item saved)
Add
Del(delect)
Save
Exit
Quit
Dial
Del
One
All
Exit
OK
Dial
Edit

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Del
Save
Exit
EDia(edit number
before dial)
Del
Dial
Exit
Exit
Quit
DND(Do not
disturb)
-Dnd
(Cancel DND)
More
FWD
(call forward)
SIP1 FWD to:
SIP2 FWD to:
SIP3 FWD to:
SIP4 FWD to:
OFF
ON
123/ALL/ABC/abc
(input mode)
Quit
SMS(Message)
New
OK
Del
Send
Exit
Quit
More
When off-hook
Dial
DIR
When dial
Dial
Del
CLR(Clear)
Exit
When called

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Ans(Answer the
call)
Deny(refuse to
answer the call)
Call state
Conf(Conference)
Clog
DIR
FWD
Retr
Xfer(Call transfer)
Xfer
Del
Bxfr
(Blind Transfer)
Exit
Hold(hold the call)
Resu
(Resume the call)
Have missed call
LCR(Call back)
Miss
(Enter missed call)
Press "Menu"
Next
OK
Quit
Menu display
01 PhoneBook
Dial
Del
OK
Exit
02 Speed Dial
Del
One
All
Exit
Edit
Del
Save
Exit
Exit

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03 Call history
Next
OK
Exit
01 Outgoing Call
Dial
Del
OK
Exit
02 Incoming Call
Dial
Del
OK
Exit
03 Missed call
Dial
Del
OK
Exit
04 Preferences
01 Do Not Disturb
Edit
Exit
02 Call Forward
Mode
Number
03 Call Waiting
04 Hotline
05 Dial Rule
01 End With *
02 Fixed Length
01 Enable Fixed
Length
02 Fixed Length
Value
03 Time Out
01 Enable Time
Out
02 Time Out Value
Network
01 Net Mode
Static
DHCP

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PPPoE
02 Static Set
01 IP
02 Netmask
03 Gateway
04 DNS
03 PPPoE Set
01 Account
02 Password
04 VLAN
06 SIP Set
01 SIP1
01 SIP Name
02 SIP Server
03 SIP Server
Port
04 SIP Number
05 SIP Account
06 SIP Password
07 SIP Register
02 SIP2
01 SIP Name
02 SIP Server
03 SIP Server
Port
04 SIP Number
05 SIP Account
06 SIP Password
07 SIP Register
03 SIP3
01 SIP Name
02 SIP Server
03 SIP Server
Port
04 SIP Number
05 SIP Account
06 SIP Password
07 SIP Register
04 SIP4
01 SIP Name
02 SIP Server
03 SIP Server
Port

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04 SIP Number
05 SIP Account
06 SIP Password
07 SIP Register
07 Phone Set
01 Screen Set
01 Contrast
02 Brightness
02 Ringer Set
01 Ringer Volume
02 Ringer Type
03 Volume Control
01 Voice Volume
02 Mic Volume
08 Product Info
01 Mac Address
02 System Version
03 Model Information
04 Network Status
05 SIP Status
06 IAX2 Status
09 System Config
01 Menu Password
02 KeyLock Password
03 Reboot System
04 Factory Reset

6th
6th、 Basic functions and operations

1、Answer the calls

When there is an incoming call, AT640 will remind user with ringing. There are 5
ways to answer the call
A、Answer by handset
Pick up the handset and talk with the caller. If you want to hang up, just put back
the handset.

B、Hand-free mode
Press the hand-free button in the phone and talk with callers by built-in
Micro-phone and Speaker. If you want to hang up, please press the hand-free

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button again.

C、Answer by earphone
Keep your earphone connected with the RJ9 earphone jack, when there is an
incoming call, press the earphone button on the IP phone and talk with the caller. If
you want to hang up, please press the earphone button again.

D、Handset to hand-free
When you are phoning with the handset and want to phone with hand-free mode,
please press the hand-free button and put down the handset.
E、Hand-free mode to handset
If you are calling under hand-free mode and want to change to speaker phone,
juts pick up the handset without press any buttons.

2、Make Call

A、Use the handset

Pickup the handset, the LCD will show the current lines (user could switch
between line1 to line4 by pressing the line button beside the LCD). User can input
the number with the keyboard and press # to send the number. When you hear the
tones of “du~~du~~” with dialed number showed on the LCD, the called’s phone is
ringing. If the called answer the call, the phone call is established and the LCD will
show the calling time and the called’s number.
B、Answer the phone under hand-free mode
Press the Speaker Phone button, the LCD will show the current lines (user could
switch between line1 and line2 by pressing the line button beside the LCD). User can
input the number with the keyboard and press # to send the number. When caller
hear the tones of “du~~du~~” with dialed number showed on the LCD, the called’s
phone is ringing. If the called answers the call, the phone call is established, and the
LCD will show the calling time and the called’s number.

C、Used phone book

a、 Pick up the handset.
b、 Press " Menu” button and use the "up" and "down" keys to enter phonebook.
c、 Press "OK" to show the total amount in telephone.
d、 Press "OK" to enter the phone list and use "up" and "down" keys to find the
contact person.
e、 When you find the certain contact person, press" OK" to show the details.
f、 Press "Edit" to edit the number or press" Dial" to call.

3、Speed dial

It’s method for the phone in standby mode to dial number immediacy.

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The method is as below：

A、Dial-up the number in standby mode
B、Push soft button “dail”, ”#”key or hang up directly to send the dial number.
C、Push soft button to save the number in telephone directory.

4、Multiple line dial-up

AT640 IP phone supports 4 Sip lines. That means user can register on4different
sip accounts simultaneity in the same IP phone. The User can choose line1 、line2
、line or line 4 to switch dial-up, System default Sip1 when dial-up.
IP Phone be called:
AT-640 maximum supports one incoming call when it is called, when the second
line calling, the LCD will show the incoming telephone number. The User can press
the “corresponding line key” indicated by LED flicker, or press soft button “ANS” to
receive the second line call, when two calls coming together, press soft button
“SWIT” to Switch.
Notice:
The phone must work with Call Waiting function when work for this feature.

5、Hang up the phone

1) Headset hang up
When use handset mode calling, put back the handset to hang up.
2) Hands free hang up
When use hands free calling, press soft button “speaker phone” to hang up.
3) Earphone Hang up
When use Earphone calling, Press the soft button “headset” to hang up.
4) Hang up one line call
When 2 lines call simultaneous, press soft button “SWIT” to choose the line
which you want to hang up, then press soft button “#” to end the call. In the
mean time, it will automatic switch to another line and continue call. Moreover,
user can redial-up or accept the second call.
Notice:
Hang up with “#” is invalidation when only one line call.

6、Call Transfer

� Blind Transfer

User A.B.C, assume B is AT-640 IP phone

1) When A Calls B and B receives
2) B presses soft button “Xfer”

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3) B dials C’s number.

4) After dialing C, B Presses soft button “Bxfr” (or wait for a few seconds after dial
number), then the call will be transfered to C.
5) When C’s phone ring, B will be hung up , the LCD on B’s Phone shows “Pls. Hang
Up”.
6) C receives the call, start to talk with A.
Notice:
SIP lines are not available for choosing when call transfer.

� Attended Transfer
User A.B.C, assume B is AT-640 Ip phone
1) When A Calls B and B receives
2) B presses soft button “Xfer” .
3) B dials C’s number and C receives to talk with B.
4) B can press "Swit" to switch the call between A and C.
5) B Presses soft button “Xfer”, then transfers the A's call to C.
6) Meanwhile The LCD on B’s phone shows “Call Ended ”.
Remarks:
To carry out this function, IP Phone must work with Call waiting and call transfer
function; meanwhile Sip server must support RFC3515.

� Alert Transfer
User A.B.C, assume B is AT-640 Ip phone
1) When A Calls B with B receives.
2) B presses soft button “Xfer” when A is calling.
3) B dials C’s number.
4) After dialing C, B directly Presses soft button “Xfer”, then transfers the call to C.
5) C receives the phone, starts to talk to A.
Remarks:
To carry out this function, IP Phone must work with Call waiting and call transfer
function; meanwhile Sip server must support RFC3515.

7、Call Hold

User can hold the current call by pressing soft button “Hold”. And by pressing
soft button “Resu” , user can get back to the previous call. In 3-way conference call
mode, user can also press this button to hold 3-way conference call, and if you press
it again, user can go back to 3-way conference mode. If hang up without exiting the
status of hold. The conversation will not be cancelled; the line is still on hold.

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8、3-Way Conference Calls

Assume B is AT-640 phone among user A,B and C.

A calls B and talks with B through VoIP.
1) B can press soft button “conf” to hold the call with A.
2) Then B inputs C’s number.
3) B presses Soft button “dial” to call to C.
4) C is on the call with B and A is on hold.
5) B presses Soft button “Conf” button to make 3-way conference call.
6)B presses soft button “spli” to end 3-way conference call and returns to the call
with A while C is holded.B can press "Swit" to switch the call between A and C.
7)B presses soft button “exit” to end all the calls.

9、Call History

AT-640 supports 100 missed calls, incoming calls and dialed calls record. When
the storage is full, the latest call will update the history. When the phone reboots or
be out of power, all the call history will be cleared.
� Missed call
1) When the LCD screen displays “(number) Missed call(s)”, press soft button
“Miss”, then the screen shows all the “Missed Call”.
2) Press navigation button to browse missed call history.
3) Choose the missed call record, press “OK” soft button to browse the specific
information of the record.
4) Press “Dial “soft button to call back it.
5) Press “Edit“soft button to edit the item and save number.
6) Press “EDia “soft button to revise the records and press soft button “dial” to call
this number.

� Incoming call
Method 1,
1) Press the "MENU" button.
2) Press the navigation button to choose “call history” and then press OK button.
3) Press the navigation button to choose “incoming call”, press soft button OK.
4) Press the navigation button to browse the incoming call record. If there is no
record, the LCD screen display “List is Empty”.
Method 2,
1) Press the left button of navigation button, you can see all the Incoming Call.
� Out coming call
Method 1,
1) Press “MENU”
2) Press up or down navigation key, and select call history and press soft button

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“OK”
3) Select “Outgoing call” through “up” or “down” key, and press soft button “OK”
4) Press up or down navigation button and check the received calls, LCD will show
“List is Empty”, if there is no received incoming call.
Method 2,
1) Press the left button of navigation button, you can see all the call history.
2) Press the left button again to switch between incoming Call, Outgoing Call and
Missed Call.

10
10、 Call pickup

Call pickup is simulated from “Pickup” function processes from IPPBX. When A call B
with no reply after ring tones, C could pick up the call from A for B by inputting the
prefix and B’s phone No.

C needed to set the dial peer with prefix code as follow

To refer *1* as the set prefix code, C could get the call from A to B by dialing *1*+B,
*1* prefix could be freely set as long as no confliction with other dialing rules. Del
Length is the digits of the prefix.

11
11、 Join call

“A” could join in the conference call，by input a prefix plus a phone No. which is
already in the conference.
A requested to set the prefix code for dial peer as follow

To refer *2* as the set prefix code， “A” could join in the conference by dial *2* plus
the call No. which is already in the conference.
*2* prefix could be freely set as long as no confliction with other dialing rules. Del
Length is the digits of the prefix.

12
12、 Redial/Unredial

In order to being efficiently to contact the busy line, A could use Redial to call B
the busy line with the prefix(for example *3*). When B is free A could get through

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the call as usual. When B is busy, A could hang up the phone while A's phone keep
checking B’s situation every 60s.When B is available, A's phone will ring. It would
call B automatically once A picking up handset.
A could dial B's number with the prefix(for example *4*) to cancel the call
before the phone automatic redialing if A is not available suddenly or don’t want to
call B anymore.

*3* is the prefix. Then A could make the redial function via dialing *3* + B's
number.
*4* is the prefix. Then A could make the unredial function via dialing *4* + B's
number.
User could set any prefix if it is compliant with present dial rule. Del Length is the
digits of the prefix.

13
13、 vport

Vport makes more flexible calling application. Eg. It could forward a call from Line 1
to one account of Line 2 after configuring forward type and number@line via web
interface. The forward could make either from Line 1 to Line 2 or Line 2 to Line 1.
But the end user may not aware the configuration being made therefore probably
the end user should be advised that it may cost with the forward function. The
forwarding could be done via either Line Key to select the line or dialing IP after
calling under server. It could be implemented by the following ways:
� Point to Point Call Forward
Make the configuration like @ip:port in the column of Forward Number. Then it
could make SIP call point to point with this IP and port in system. User could select
forward type accordingly.

� Point to Point Blind Transfer

Transfer the call via dialing IP directly.
Call Forward, Call Transfer (Blind Transfer/ Attended Transfer) in different Line.
Make the configuration like sip: username@n in the column of Forward Number.
Then system would select Line N and make call accordingly.
SIP Line (eg:0/1/2. Or 0.0.0.0/0.0.0.1/0.0.0.2/255.255.255.255 which is
compliant with former configuration).
Call Forward, Call Transfer (Blind Transfer/ Attended Transfer) between SIP Line
and Point to Pint.
It is compliant for the Call Forward, Call Transfer (Blind Transfer/ Attended
Transfer) between SIP Line and Point to Pint.

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14、 Click to dial

When User A accesses web interface and calls User B via clicking a link which is
direct to B, IP Phone of User A would ring and calls B automatically once A picking up
handset(need server support).

15、 SMS function

15

� Create new SMS

1) press MORE (soft button 4)
2) press SMS(soft button 2）
3) press New(soft button 1)
4) Edit SMS context and you can switch the input method by press # such as
ABC(capital letters) ，abc ( lower case letters) , 123 ( number).
5) When the edit is done , press Send(soft button 2）and input receiver’s
phone number.
A、press Sear(soft button 1) to find the contact person in phonebook
B、directly input receiver’s phone number
C、Use P2P method , input # + IP address ( press * twice to input #)
For example if you send the SMS to the phone with IP address of
192.168.1.88, you will press **192*168*1*88
After inputing receiver's address, press Send(soft button 2) to send out your
message.
� SMS Check new SMS
When there is a new SMS, LCD will show New Message(S)
6) Press More(soft button 4).
7) Press MS(soft button 2） , LCD will display Number New Number old.
8) If there is a new SMS and 2 old SMS, LCD will display 1 New 2 Old.
9) Press OK (soft button2) to enter SMS list , if it’s unread , there will be a NEW
before it , or else it has been read.
10) Press up and down key in navigation keyboard to select the message and
press OK(soft button2) to read it.
11) If you want to delete the SMS , just press Del(soft button 1) after you select
it.
12) If you want to reply the message,just press Repl(soft button 2) after you
select it.

Notice
Notice：
：
In SMS list, you can press quit (soft button) to go to the upper menu
Dial means dial to call sender directly when you are reading his SMS
Edia means call the sender after edit his number .
Edit means editing the SMS context.

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16、 Default Password

There are 2 models to set the authority of web accessing and command line: Guest
model and Admin model. User could view and configure all items in Admin mode.
While guest couldn’t change the SIP (1-2) and IAX2 configuration as well as server
address and port but only access and view the information. User would enter
different mode after input different user name and password:
� Web interface:
� Guest Model
Username：guest Password：guest
� Admin Model：
Username：admin Password：admin

� Keypad password: 123

17
17、 Phone’’s IP
Check the Phone

Press the up or down navigation button to check the phone’s current IP address.

7th
7th、 Web settings

Enter AT-640 IP addresses in the web browser to go to the log on page, and key in
the username and password to access AT-640 setting page.
Default username and password is:
Administrator: Username: admin password: admin
User: Username: guest guest
password:guest

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1、Current state

This page shows the IP phone working status.

The network part shows the connection status of WAN and LAN.
Phone Number part shows the phone number and register status for Line1、Line2
and IAX2.
The Version shows the current firmware version.

2、Network

2.1. Wan Config

There are 3 ways to connect to the internet DHCP, Static and PPPoE, please
choose one according to your own situation.
A、DHCP，the IP phone will get IP address from DHCP server , you do not have
to fill in the date of IP address , net mask etc , just choose DHCP and submit . Please
refer to the below picture.

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Parameters
Parameters：
� Active IP： IP phone’s address.
� Current Net mask：network net mask.
� MAC Address：MAC of IP phone.
� Current Gateway：the IP address of the router.

B、If your ISP provide you with the fixed IP address, please choose static and fill
in the correct information of IP Address、 Net mask、Gateway、Primary DNS etc.
If you do not know it please refer to your ISP provider or network management
stuff. The reference picture is as below.

Parameters
Parameters：
� Static IP Address: fixed IP address.
� Net mask: LAN net mask.
� Gateway: Gateway IP address.
� DNS Domain: input DNS domain name if it’s provided.
� Primary DNS: Primary DNS address.
� Alter DNS: Alternative DNS address.

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C、when you use PPPoE to get IP address，please select “PPPoE”，and input ADSL
account information as below picture：

Parameters：
PPPoE Server: sever name, if the ITSP have no special requirements, please keep
"ANY" as default.
Username: ADSL account username.
Password: ADSL account password.

Attention：
1）After configuration setting please click “Apply” to apply changes.
2 ） If the IP address is changed after effecting the configuration change , the
webpage will lose response former address, so you must get to the webpage with
new address.
3）If the LAN IP address is happened to be the same as WAN IP which is allocated
from DHCP server. The LAN IP address will be changed automatically by adding 1 at
the last digital.

2.2. LAN Config

Parameter
Parameter：
� LAN IP：config LAN static IP.

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� Net mask: LAN net mask.

� DHCP Service: enable LAN DHCP Server , need to reboot to make it available.
� NAT: Network Address Translation.
� Bridge Mode: Select Bridge Mode or not: If you select Bridge Mode, the phone
will no longer set IP address for LAN physical port，LAN and WAN will join in the
same network. Click “Apply”, the phone will reboot.

3、 VoIP

3.1. SIP1/SIP2/SIP3/SIP4

� Register Status: SIP server registration status, if succeed display Registered，

or else display Unregistered.
� Server name: SIP server name , if no special requirements just keep it as
blank.
� Server Address: SIP server address , support both IP address and domain
name.
� Server Port: SIP server port , default is 5060.
� Account Name：SIP account name.
� Phone Number：SIP account phone number, if leave it as blank , no registration
information will be sent out.
� Display Name：Show the display name that you want to display on the phone of
callee. Support number and letter input.
� Proxy Server Address：Normally the Proxy server is the same as SIP server. If
they are different then fill in the correct information that provided by ISP.
� Proxy Server Port: Set your SIP server port.
� Proxy Username: Input your SIP register account name.

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� Proxy Password: Input your SIP register password.

� Domain Realm: config SIP local domain. If the server does not have special
requirements for the local domain of SIP terminal, the local domain can be the
same as SIP server domain. The user can also leave it as blank; the system will
take SIP server domain as the domain realm.
� Enable Register: Enable or disable registration .
Advanced SIP setting

� Register Expire Time: register expire time, default is 60 seconds. AT-640 will
auto configure this expire time to the server recommended setting if it is
different from the SIP server.
� Auto Detect Server Interval: Set examining interval of the server, default is 60
seconds.
� User Agent: Set the user agent if have, the default is VoIP Phone 1.0.
� Signal Key: Signal encryption Key.
� Media Key: voice stream encryption Key.
� Local Port: Local SIP signal port，default as 5060.
� Hotline Number: Set hot line number of each line.
� MWI Number: Set SIP1 voicemail Number.
� Enable Keep Authentication: Enable/Disable Keep Authentication System will
take the last authentication field which is passed the authentication by server to

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the request packet. It will decrease the server’s repeat authorization work, if it
is enable.
� Auto Detect Server: Enable/Disable keeps NAT of SIP alive. If some server
refuse to register with too short interval time, and has no packets sending to
device in private network to keep NAT alive, user could set this function ON. It
need set the keep alive interval time less than the NAT server’s.
� Enable Via rport: Enable/Disable system to support RFC3581. Via rport is
special way to realize SIP NAT.
� Enable PRACK: Enable or disable SIP PRACK function, suggest use the default
config.
� Long Contact: Set more parameters in contact field.
� Click to Talk: Set click to Talk (need practical software support).
� Ban Anonymous Call: Set to ban Anonymous Call.
� Dial Without Register: Set call out by proxy without registration.
� Enable Strict Proxy: Support the special SIP server-when phone receives the
packets sent from server， phone will use the source IP address, not the address
in via field.
� Enable Conference Num: conference ID.
� Forward Type: Select call forward mode, the default is Off .
� Off：Close down calling forward.
� Busy：If the phone is busy, incoming calls will be forwarded to the appointed
phone.
� No answer ： If there is no answer, incoming calls will be forwarded to the
appointed phone.
� Always：Incoming calls will be forwarded to the appoint phone directly.
The phone will prompt the incoming while doing forward.
� Forward Phone Number: Appoint your forward phone number.
� Server Type: Select the special type of server which is encrypted, or has some
unique requirements or call flows.
� DTMF Mode: Select DTMF sending mode, there are three modes:
� DTMF_RELAY
� DTMF_RFC2833
� DTMF_SIP_INFO
Different VoIP Service providers may provide different modes.
� RFC Protocol Edition: Select SIP protocol version to adapt for the SIP server
which uses the same version as you select. For example, if the server is
CISCO5300, you need to change to RFC2543; else phone may not cancel call
normally. System uses RFC3261 as default.
� Transport Protocol: Set transport protocols, TCP or UDP.
� Subscribe Expire Time: Overtime of resending subscribe packet. Suggest using
the default config.
� Conference Number: config certain Conference call number.
� Signal Encode: enable signal encryption.
� Rtp Encode: enable voice data encryption.

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� Enable Session Timer: enable rfc4028 to refresh the SIP sessions.

� Answer With Single Codec: only answer the call with a certain Codec.
� Auto TCP: enable TCP transmission protocol when the length of message exceed
1300 byte.
� Enable URI Convert: convert # into %23 when sending URI.
� Enable Display name Quote: Set to make quotation mark to display name as the
phone sends out signal, in order to be compatible with server.
� Enable GRUU: Set to support GRUU.
� Enable Subscribe: Enable Subscribe: Overtime of resending subscribe packet.
Suggest using the default config.

3.2. Iax2 Config

Above is the IAX server configuration page

� IAX Server Addr: Register address of public IAX server.
� IAX Server Port: Register port of public IAX server，default port is 4569.
� Account Name: Username of your SIP account (Always the same as the
phone number).
� Account Password: Password of your IAX account.
� Local port: Signal port of local, default port is 4569.
� Phone Number: Phone number of your IAX account.
� Voice mail number: If the IAX support voice mail, but your username of the

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voice mail is letters which you cannot input with the ATA , then you use the
number to stand for your username.
� Voice mail text: if IAX support voice mail, config the domain name of your mail
box here.
� Echo test number: If the platform support echo test , and the number is test
form , the config the test number to replace the text format The echo test is to
test the error status of terminals and platform.
� Echo test text: echo test number in text format.
� Refresh time: IAX refresh time.
� Enable Register: enable or disable register.
� Enable G.729: Using G.729 speech coding mandatory consultations.

4、 Advanced

4.1. DHCP Server

� DHCP Leased Table: IP-MAC mapping table. If the LAN port of the phone
connects to a device, this table will show the IP and MAC address of this device.
� Leased IP Address: the IP address which is assigned.
� Client Hardware Address: the IP address assigned and the MAC opposite of IP.
DHCP Lease Table Setting:
� Lease Table Name: Lease table name.
� Lease Time: DHCP server lease time.
� Start IP: Start IP of lease table.
� End IP: End IP of lease table. Network device connecting to the AT640 LAN port
can dynamic obtain the IP in the range between start IP and end IP.
� Net mask: Net mask of lease table.
� Gateway: Default gateway of lease table.

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� DNS: default DNS server of lease table.

� Press “add” to apply, will added DHCP lease table.
� Lease Table Name: Select name of lease table, click the Delete button will
delete the selected lease table from DHCP lease table.
� DNS Relay: Select DNS Relay, the default is enable. Click the Apply button to
become effective.
� DHCP Lease Table: Shows the DHCP Lease Table ， the unit of Lease time is
Minute.
Notice:
1）The size of lease table cannot be larger than the quantity of C network IP address.
We recommend you to use the default lease table and not modify it.
2）If you modifies the DHCP lease table, you need save the configuration and reboot.

4.2. NAT

NAT is abbreviated from Net Address Translation; it’s a protocol responsible for IP
address translation. In other word, it is responsible for transforming IP and port of
private network to public, also is the IP address mapping which we usually say.

DMZ config：
In order to make some intranet equipments support better service for extranet, and
make internal network security more effectively, these equipments open to extranet
need be separated from the other equipments not open to extranet by the
corresponding isolation method according to different demands. We can provide the
different security level protection in terms of the different resources by building a
DMZ region which can provide the network level protection for the equipments
environment, reduce the risk which is caused by providing service to distrust
customer, and is the best position to put public information The following chart
describes the network access control of DMZ.

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The setting page as below:

� IPSec ALG: It is an encryption technology. Select it to enable IPSec ALG, the

default is enable.

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� FTP ALG: FTP is a service of connection layer which can transform intranet IP
into extranet IP when intranet IP is sending out packet. Select it to enable FTP
ALG, the default is enabling.
� PPTP ALG: Select it enable PPTP ALG, the default is enable.

Shows the NAT TCP mapping table

Shows the NAT UDP mapping table.

NAT Table Option:
� Transfer Type：Select the NAT mapping protocol style, TCP or UDP.
� Inside IP: Set the IP address of device which is connected to LAN interface to do
NAT mapping.
� Inside Port: Set the LAN port of the NAT mapping.
� Outside Port: Set the WAN port of the NAT mapping.
Notice: After finish setting, click the Add button to add new mapping table.
click the Delete button to delete the selected mapping table.

� DMZ Table: Shows the outside WAN port IP address and the inside LAN port IP
address.
� Outside IP：Set the outside wan port IP address of DMZ.
� Inside IP：Set the inside LAN port IP address of DMZ.
Click the "Add" button to add new table; click the "Delete" button to delete the
selected mapping table.
Notice: 10M/100M adaptive means the network card, and other equipment physical
consultations speed, testing speed under bridge mode near to 100M, in order to
ensure the quality of voice and communications real-time performance, we made
some sacrifices of NAT under the transmission performance. Transmit with full
capability only when system is idle, so cannot guarantee that the transmission
speed reach to 100M.

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4.3. Net Service

� HTTP Port: set web browser port, the default is 80 port，if you want to enhance
system safety，you'd better change it into non-80 standard port； Example: The
IP address is 192.168.10.88. and the port value is 6090, the accessing address
is http://192.168.10.88:6090
� Telnet Port:Set Telnet Port, the default is 23. You can change the value into
others. Example: The IP address is 192.168.1.88. the telnet port value is 6023,
the accessing address is telnet 192.168.1.88:6023
� RTP Initial Port: Set the RTP Initial Port. It is dynamic allocation.
� RTP Port Quantity: Set the maximum quantity of RTP Port, the default is 200.
Notice:
1）You need save the configuration and reboot the phone after set this page.
2）If you modify the port of Telnet and HTTP, you would better set the value more
than 1024 because the port value less than 1024 is system port reserved.
3）if you set 0 for the HTTP port, it will disable HTTP service.

4.4. Firewall

� in_access enable： Select it to Enable in_ access rule.

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� out_access enable： Select it to Enable out_ access rule.

� Firewall Input Rule Table: Firewall input rule, as the picture config is deny
192.168.1.2 ping 192.168.10.2, but ping 192.168.10.0/24 beside
192.168.10.3 is ok.
� Firewall Output Rule Table: Firewall output rule, as the picture config is the
phone ping 192.168.1.70 was deny.

� Input/output: Specify current adding rule by selecting input rule or output rule.
� Deny/Permit: Specify current adding rule by selecting Deny rule or Permit rule.
� Protocol Type: Filter protocol type. You can select TCP, UDP, ICMP, or IP.
� Port Range: Set the filter Port range.
� Src Addr: Set source address. It can be single IP address, network address,
complete address 0.0.0.0, or network address similar to *.*.*.0.
� Dest Addr: Set the destination address. It can be IP address, network address,
complete address 0.0.0.0, or network address similar to *.*.*.*.
� Src Mask: Set the source address’ mask. For example, 255.255.255.255 means
just point to one host; 255.255.255.0 means point to a network which network
ID is C type.
� Des Mask:: Set the destination address’ mask. For example, 255.255.255.255
means just point to one host; if set to 255.255.255.0 means point to a network
which network ID is C type.

4.5. Qos

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� VLAN Enable：Before select it to enable VLAN, you need enable Bridge mode in
LAN config.
� VLAN ID Check Enable：Enable VLAN ID check by selecting it. After enable VLAN
ID check, if VLAN ID of a data package is not the same with the phone’s or a
data package do not have VLAN ID, the data package will be discarded.
� Voice/Data VLAN differentiated：After enable VLAN, system will set packets with
different type of VLAN ID. Undifferentiated means after using VLAN, both voip
packets and other data packets will use the voice VLAN ID; tag differentiated
means after using VLAN, VoIP(signal and voice) packets will add voice VLAN ID,
and other data packets will add data VLAN ID; data untagged means after using
VLAN, only VoIP packets will add voice VLAN ID. Other data packets will not use
VLAN.
� DiffServ Enable：Select it or not to Enable or disable DiffServ.
� DiffServ Value：Set DiffServ value, the common value is 0x00.
� Voice 802.1P Priority：Specify 802.1P Priority of voice/signal data package.
� Data 802.1P Priority：Set 802.1p of data VLAN. Non-voip data (such as http,
telnet, ping etc) will use this value to set VLAN package.
� Voice VLAN ID：Set VLAN ID of voice/signal data package.
� Data VLAN ID：Set 802.1q of data VLAN ID. Non-VoIP data (such as http, telnet,
ping etc) will use this value to set VLAN package.
Notice
Notice：
1）Enable VLAN, if set Voice and Data VLAN differentiated as Undifferentiated, all
packets will use the Voice VLAN ID as the tag.
2) Enable VLAN, if set Voice and Data VLAN differentiated as tag differentiated and
disable the DiffServ, then system will not distinguish the voice and data, all packets
will use the Voice VLAN ID as the tag.
3) Enable VLAN, if set Voice and Data VLAN differentiated as tag differentiated and
enable the DiffServ, then system will distinguish the voice and data and add the
VLAN ID each other.
4) Enable VLAN, if set Voice and Data VLAN differentiated as date untagged, then
the packet of the signal and voice will use the voice VLAN ID as the tag, but the data
packets will not take the VLAN tag.
5,if disable the VLAN, regardless to set the voice and data VLAN differentiated or not,
all packets will not take the VLAN tag; if enable the DiffServ, all packets will only
take the DiffServ value.
6) One must to notice, enable the VLAN ID check enable that is default, if enable
� Must to notice, VLAN ID check Enable feature is default enable, if enable it, The
phone will match the VLAN ID strictly, When others’ VLAN ID mismatch with IP
Phone, the packets will discard, Contrarily, the phone will accept the packets
with the distinct VLAN ID.
� You must set the IP with static mode when you set VLAN, otherwise can’t obtain
the IP in the VLAN and also cannot dial with point to point

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4.6. Digital Map

Digit map is a set of rules to determine when the user has finished dialing.
AT640 support below digital map:
� End With “#”: Use # as the end of dialing.
� Fixed Length: The call will be sent out automatically when the length of the
number you dial reaches the fixed one. For example if you set number of 11
here, when you dial 11 digits the call will be sent out immediately.
� Timeout: Specify the timeout of the last dial digit. The call will be sent after
timeout.
� Prefix: User define digital map:
� [ ] represents the range of digit, can be a range such as [1-4], or use comma
such as [1,3,5], or use a list such as [234]
� x represents any one digit between 0~9
� Tn represents the last digit timeout. n represents the time from 0~9 second, it
is necessary. Tn must be the last two digit in the entry. If Tn is not included in
the entry, we use T0 as default, it means system will sent the number
immediately if the number matches the entry.
� Example:
� [1-8]xxx All number from 1000 to 89999 will be sent immediately.
� 9xxxxxxx 8 digits numbers begin with 9 will be sent immediately.
� 911 Number 911 will be sent will be immediately.
� 88xT4 3 digits numbers begin with 88with be sent after 4s.
� 6611x.T4 holds four seconds send out if the number begins 6611 and
five digits.

Notice ： The above configuration can exist at the same time. For example you
enable # as the signal of sending the call while set fixed length of 11. Either you
press # before the number reach 11 or dial 11 digital can send out the call.

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4.7. Stun

� STUN NAT Transverse：STUN NAT Transverse status true or false.

� STUN Server Addr: configure stun server address.
� STUN Server Port: configure stun server port default 3478.
� STUN Effect Time: stun detect NAT type interval time .If NAT found a link
inactive for a certain time , it will close the link so you need to send a packet
within a interval tome to keep the link alive.
� Local SIP Port: config local SIP port , default as 5060Use Stun：enable/disable
SIP STUN.
Attention：
SIP STUN is used for NAT transverse. When you config STUN server’s address
and port (default 3478) and enable it, then you can use the normal SIP server
to make the IP phone transverse NAT.

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4.8. Call Service

� No Answer Time: no answer call forward time setting.

� No Disturb: DND, do not disturb, when there is an incoming call , the caller will
get the message that this line is not available , but you it has no affection when
you make outgoing call.
� Ban Outgoing: Enable this to ban outgoing calls.
� Enable Call Transfer: Enable Call Transfer by selecting it.
� Enable Call Waiting: Enable Call Waiting by selecting it.
� Enable Three Way Call: 3 way conference call.
� Accept Any Call:If select it, the phone will accept the call even if the called
number is not belong to the phone.
� Auto Answer:If select it, the phone will auto answer when there is an incoming
call.
� P2P IP Prefix: Set Prefix in peer to peer IP call. For example: what you want to
dial is 192.168.1.119, If you define P2P IP Prefix as 192.168.1., you dial only
#119 to reach 192.168.1.119. Default is “.”. If there is no “.” Set, it means to
disable dialing IP.
� Black List: Set Add/Delete Black list, incoming call in these phone numbers will
be refused.
It support below rules:
� You add a certain number in it , when this number call you , it will be refused.
� Use “x” to represent any number. For example , 4xx means any incoming call
with 3 digital and the first digital is 4 , will be refused.
� DOT (..) means matching any arbitrary number digit. for example, any
number with prefix 6 will be forbidden to dialed out. Any digital call with a
certain head number, For example 6. means any incoming number with the 6 as
the first number will be refused.
� if user wants to allow a number or a series of number incoming, he may add

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the number(s) to the list as the white list rule. the configuration rule is –number,
for the settings as below.

-7049 means any incoming number is forbidden except 7049

Notice: End with DOT (.) when set up the white list
� Limit List: Set Add/Delete Limit List. Please input the prefix of those phone
numbers which you forbid the phone to dial out. For example, if you want to
forbid those phones of 001 as prefix to be dialed out, you need input 001 in the
blank of limit list, and then you cannot dial out any phone number whose prefix
is 001. x and . are wildcard. x means matching any single digit. for example,
4xxx expresses any number with prefix 4 which length is 4 will be forbidden to
dialed out . Means matching any arbitrary number digit. For example, 6.
expresses any number with prefix 6 will be forbidden to dialed out.

4.9. MMI Filter

User could make some device own IP, which is pre-specified, access to the MMI
of the phone to config and manage the phone.
Add or delete the IP address segments that access to the phone. Set initial IP
address in the Start IP column, Set end IP address in the End IP column, and click
Add to add this IP segment. You can also click Delete to delete the selected IP
segment.
Notice: Do not set your visiting IP outside the MMI filter range, otherwise, you
cannot logon through the web.

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4.10.Audio Settings

� First Codec：The fist preferential DSP codec:G.711A/u, G722, G.723, G.726-32,

G.729.
� Second Codec ： The second preferential DSP codec:G.711A/u, G722, G.723,
G.726-32, G.729.
� Third Codec ： The third preferential DSP codec:G.711A/u, G722, G.723,
G.726-32, G.729.
� Forth Codec: The Forth preferential DSP codec:G.711A/u, G722, G.723,
G.726-32, G.729.
� Fifth Codec: The fifth preferential DSP codec:G.711A/u, G722, G.723, G.726-32,
G.729.
� Sixth Codec: The sixth preferential DSP codec:G.711A/u, G722, G.723,
G.726-32, G.729.
� Input Volume：Specify Input (MIC) Volume grade.
� Output Volume：Specify Output (receiver) Volume grade.
� Hands free Volume：Specify Hands free Volume grade.
� Ring Volume：Specify Ring Volume grade .
� G729 Payload Length：Set G729 Payload Length.
� Signal Standard：Select Signal Standard.
� G722 Timestamps: 160/20ms or 320/20ms is available.
� G723 Bit Rate: 5.3kb/s or 6.3kb/s is available.
� Default Ring Type: Select signal standard.
� VAD：Select it or not to enable or disable VAD. If enable VAD, G729 Payload
length could not be set over 20ms.
� Handdown Time: Set the time which the phone hang up automatically after call
ended when use handfree mode.

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4.11. VPN

this page is VPN setting page , the IP phone support the VPN with UDP and L2TP
protocol .The parameters is as below.
� VPN IP： After VPN registered successfully, VPN server will give an IP aggress to
the terminal. If there is a IP address shown on terminal (except for 0.0.0.0), it
means your VPN has registered.
� L2TP
� VPN Server Addr： Register to the address of VPN server
� VPN User Name： L2TP VPN username
� VPN Password： L2TP VPN password
� L2TP: use the L2TP to visit VPN
� Enable VPN: Enable the VPN server, you must choose UDP or L2TP type in
advance

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5、Dial Peer

This functionality offers you more flexible dial rule, you can refer to the following
content to know how to use this dial rule. When you want to dial an IP address, the
entry of IP addresses is very cumbersome, but by this functionality, you can set
number 179 to replace 192.168.1.179 here.
When you want to dial a long distance call to China, you need dial an country code
86 before local phone number, but you can also dial number 0 instead of 86 after we
make a setting according to this dial rule. For example, you want to dial
8675583018619, but you need dial only 075583018619 to realize your long
distance call after you make this setting.

AT640 provide flexible dial rule, with different dial-rule configure, user can easily
implement the following function:
----Replace, delete or add prefix of the dial number
----Make direct IP to IP call
----Place the call to different servers according the prefix
You can click “Add” to add a new dial rule. Below is the detail setting of the dial-rule:

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� Phone Number: The Number suit for this dial rule, can be set as full match or
prefix match. Full match means that if the number user dialed is completely the
same as this number, the call will use this dial-rule. Prefix match means that if
prefix of the number that the user dials is the same as the prefix, the call will use
this dial-rule, to distinguish from the full match case, you need to add “T” after
the prefix number in the phone number setting.
� Call Mode: support SIP.
� Destination (optional): call destination, can be IP or domain. Default is 0.0.0.0;
in this case the call will be routed to the Public SIP server. If you set the
destination to 255.255.255.255, then the call will be routed to the private SIP
server. Also you can key other address here to make direct IP calls.
� Port (optional): Configure the port of the destination, default is 5060 in SIP
� Alias (optional): Set up the Alias. We support four Alias as below. Alias need to
co-work with the Del Length.
� add:xxx, add prefix to the phone number, can set to reduce the dial length.
� all: xxx, replace the phone number with the xxx, can use as speed dial function.
� Del, delete the first N numbers. N is set in the Del Length.
� rep:xxx，replace the first N numbers. N is set in the Del Length. For Example:
Use wants to place a call 8610-62281493, then you can set the phone number in the
dial rule as 010T, and set the Alias as rep:8610, and set the Del Length to 3. Then
all calls begin with 010 will be changed to 8610 xxxxxxxx.
� Suffix (optional): Configure suffix, show no suffix if not set.

Instance description as picture:

� 2T rule: if the call starts with 2, the first 2 will be deleted, and the rest number
with be sent to IAX2 Server.
� 33 rule: Dial 33 and will send 83018618 to your server. Used as speed dial
function.
� 0T rule: If the calls are begin with 0, the first 0 will be replacing by 86. Mean that
if you dial 075583018049 and AT640 will send 8675583018049 to your server.
� 179 rule: when you dial 179, the call with send to 192.168.1.179, suit for LAN
application without set up a sip server.

You can also Modify and Delete the existed Dial Peer.

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6、Config Manage

� Save Config: you can save all changes of configurations. Click the Save button,
all changes of configuration will be saved, and be effective immediately.
� Backup Config: Right clicks on “Right click here…” and select “Save Target As….”
then you will save the config file in .txt format .
� Clear Config: user can restore factory default configuration and reboot the
phone. If you login as Admin, the phone will reset all configurations and restore
factory default; if you login as Guest, the phone will reset all configurations
except for VoIP accounts (SIP1、SIP2 and IAX2) and version number.

7、Update

7.1. Web Update

Click the browse button, find out the config file saved before or provided by
manufacturer, download it to the phone directly, press “Update” to save. You can

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also update downloaded update file, logo picture, ring, mmiset file by web.

7.2. FTP/TFTP Update

� Server: FTP/TFTP server address. It can be the format of IP address such as

192.168.1.1 or domain such as ftp.domain.com Meanwhile , it support sub
directory such as 192.168.1.1/ftp/config/ or ftp.domain.com/ftp/config.
� Username: FTP user name (TFTP no need).
� Password: FTP password (TFTP no need).
� File name: the firmware or configuration file name that IP phone will search for
in the server , if leave it as blank the IP phone with search the file with the name
of its MAC such as 000102030405.

Notice
Notice： Users can revise the exported config file by themselves and import the
config file with only modules, for example if there is the SIP setting page in the
config file , the IP phone will only change SIP setting after import this file and
leave other setting as not changed.

� Type: upgrading type

� Application update: update firmware.
� Config file export: export the current configuration to a FTP/TFTP server.
� Config fie import: import configuration file from a FTP/TFTP server.
� Protocol: choose server type FTP or TFTP.

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7.3. Auto Provisioning

� Current Version: the system will display the current version number. need to
modify the version number in config file bigger than this number before auto
provision update.
� Server Address: FTP/TFTP server address.
� Username: FTP server user name.
� Password: FTP server password.
� Config File Name: The name of configuration file. Normally users leave it as
blank the IP phone search for the file with the name same as its MAC in the
server.
� Config Encrypt Key: The encrypt key of confirmation file.
� Protocol Type: The protocol type that used for upgrading.:FTP,TFTP or Http.
� Update Interval Time: The interval time that the terminals search for new
configuration file , counted in hour.
� Update Mode: auto provision mode;
A、Disable: not auto update，
B、Update after reboot: auto update after reboot，
C、Update at time interval: auto update after a certain time

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8、System Manage

8.1. Account Manage

Users can add new account or delete and change existing account
Set Menu Password: Set menu of keypad password, default is “123”
Set KeyboardLock: The default password is “123”. It will take effect when you
enable the keyboard lock. The default setting is unlock, if you press any key at this
status, the system will remind you to input password

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Set Backlight Timeout: Set backlight time out, if IP Phone has not press any
operation to active within the settings value, the backlight will off.
Set Greeting Message: set the Greeting message on the LCD, default is VOIP
PHONE.
� User Name：set new account name
� User Level：set new account level；root can read and change setting，general can
only read
� Password：config password for new account
� Confirm：double confirm password
If you want to make change on existing account , select the account an click
【 Modify
Modify】 or 【 Delete
Delete】 . General account can only modify or delete general
account
� Keyboard Password：config password that you use keyboard to access the menu
, must be in number.

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8.2. Phone Book

� Phonebook Table: shows phonebook detailed information

� Add Phone Book: add a new record in phonebook
� Name: nick name of a number ，when the call of this number comes in the LCD
will show the name
� Number: phone number
� Ring Type: set different ring for different person
If you want to make change on existing account , select the account an click 【Modify
】or 【Delete】. General account can only modify or delete general account

Notice：Maximum records of phone book is 500pcs

8.3. Multi Line Set

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Multi line function is one of SIP line is busying, but other lines can get the call when
have new calls with the line.
The settings as below:
1, go to System Manage Multi Line set page,
2, set the F1/F2/F3/F4 to Multiple line and set the same SIP:name, such as
SIP1:8047 or SIP2:8047 or IAX2:6008

The flow as below:

1,A call AT640 as 8047 and calling.
2,B call AT640 as 8047
3,LED of Line 2 is flicker
4,press Line2 or answer of softkey can call from B.

The one of 4 lines can also be configured as IAX2 line

1.Choose line (for example line2) ,set F2 to Line
2.Set SIP2:name as IAX2:name

8.4. Function Key Set

There are 9 function key on the phone, and be expanded to 29 with expander
board.There are 5 types of the key:

1.NONE: do not use this key

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2.Memory Key: set Number@Line/Subtype for the key,the number will be send out
if you press the key.
� Number@Line/b: BLF (need server support). You can see the status of the blf
number you set.When the key is green,means the number is free now. When
the key turns red and blink,means the number is ringing. When the key turns
red for steady,means the number is in a conversation.
� Number@Line/m: MWI,like the voice mail key.The key will blink if there is a new
voicemail.
� Number@Line/p: Presence(need server support).You can see the register
status of the number.
� Number@Line/f: Speed Dial
� Number@Line/i: Push to Talk(need server support)

3.Line:
� SIP1: use sip1 to call
� SIP2: use sip2 to call
� SIP3: use sip3 to call
� SIP4: use sip4 to call
� IAX2: use iax2 to call

4.Key Event:
� F_PBOOK: Phone Book
� F_REDAIL: Redail
� F_B_TRANSFER: Blind Transfer
� F_PICKUP: Pickup(need server support)
� F_JOIN: Joincall(need server support)
� F_AUTOREDIAL: Auto redial(need server support)
� F_UNAUTOREDIAL: Cancle Auto redial(need server support)
� F_DND: Do not Disturb
� F_MWI: MWI(Message Waiting Indication) only for Sip1
� F_CFWD: Call forward(always), press down key to switch between sip1, sip2,
sip3 and sip4
� F_CALLERS: Call history including Missed Call, Incoming Call.
� F_MEMO: Memorandum

5.DTMF: After set a number, if you press the key, the number will be send out.

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8.5. Syslog Config

Syslog is a protocol which is used to record the log messages with client/server
mechanism.
Syslog server receives the messages from clients, and classifies them based on
priority and type. Then these messages will be written into log by some rules which
administrator can configure. This is a better way for log management.8 levels in
debug information: Level 0---emergency: This is highest default debug info level.
You system can not work.
Level 1---alert: Your system has deadly problem.
Level 2---critical: Your system has serious problem.
Level 3---error: The error will affect your system working.
Level 4---warning: There are some potential dangers. But your system can work.
Level 5---notice: Your system works well in special condition, but you need to check
its working environment and parameter.
Level 6---info: the daily debugging info.
Level 7---debug: the lowest debug info. Professional debugging info from R&D
person.
At present, the lowest level of debug information send to Syslog is info, debug level
only can be displayed on telnet.
The items describe:
� Server IP： Syslog server IP address
� Server Port：Syslog server port
� MGR Log Level：config MGR log level
� SIP Log Level：config SIPlog level
� IAX2 Log Level：config IAX2log level
� Enable Syslog：Enable/Disable Syslog

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8.6. Time Set

� Server: type the IP address of time server

� Timezone: select correct time zone in list box
� Timeout: longest response time for SNTP
� Daylight Timeset: daylight setting through manual
� Manual Timeset: Time setting through manual
� Enable Daylight: Daylight saving time

You can also set the time manually.

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8.7. Call Log

� Start Time：Display starts time of the outgoing record.

� Last Time：Display conversation time of the outgoing record.
� Called Number：Display the account/protocol/line of the outgoing record.
Notice:
It will cover existing automatically if the call log table has the new record.Call log will
be cleared after phone reboot.

8.8. Language Set

Language Set: Set the language of phone, English is default. Because we use 14px
font on LCD so the Chinese and Korean language are not supported but only can be
supported on web. The default language is English, if you need other language
support; please feel free to contact our sales.

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8.9. Logout

Log out the configuration mode. If you want to re-configuration the phone,
need to input the user and password to login again.

8.10. Reboot
8.10.Reboot

Reboot IP phone, some settings need reboot to make it works. Please always
save config before reboot, otherwise the settings will return to previous settings.

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