ATA V600 User manual

User
Manual

1
Contents
W e l c o m e … … … … … … … … … … … … … … … … … … … . . 3
CHAPTER 1 Installation………………………..……………………………….4
CHAPTER 2 Production Summarize……………………..…………………5
I、Key Features……………………………………………………………………….5
II、Technical Parameter……………………….………………….………..…………5
CHAPTER 3 Basic Manipulation………..…….………………..…………….6
I、K e y s t r o k e s A n d V o i c e
Prompt…………………………..………..…….……………6
II、Make a Call……………………………………………..…………….…………….8
i、How to Make a PSTN Call ……..………………………………………………8
i i 、H o w t o M a k e a V o I P

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C a l l … … … … . … … … … … … … . . … … … … … … … . … … … 8
iii、Direct IP Call……………..…………………………………………………….9
III、Call Features ………………….………………………..…………..……………11
IV、Call Function………..……………….………………………………………….12
i、Blind Transfer …………………….……………………………………………12
ii、Attended Transfer …………….……………………………………………13
iii、Conference Call ……………………………………………………….………13
iv 、VoIP-to-PSTN
Calls………….………………………………………………….14
v、PSTN-to-VoIP Calls……….………………..………………………..………….14
vi、Route Calls to PSTN………………………………………………….…………14
vii、Fax ……………………………………………………………………..………15
viii、LED Light……………..……………………………………..………………15
CHAPTER 4 Configuration guide ……..…………………..………………16
I、Configuring Wan IP Through Voice Prompt……………………….…………16
i、Dhcp Mode….. ……………………………………………………………16
ii、Static IP Mode …….…………………………………………………………17
II、Configuring With Web Browser ……………..…….…………………………17
i、Access the Web Configuration Menu ….…..……………….………………18
ii、End user Configuration ………………………….…………………………18
III、Status……………………………..…………………………………….……...18
IV、Basic Optino…….……………….…………………….………………….……18
i 、L a n S e t t i ng s … … … … …… … … … … …. … … … … …… … … … …. 1 8
ii 、Wan Settings ………………………….….………………………….19
iii、Nat Settings……………………………….….………………………………21
iv、Other Settings……..……………………………………………………………22
v、Call Settings(V611Fxo port)………..……..……………………………………22
V、Super Options……..…….…………………………………………………..24
i、SIP Settings……………………………………………….………….24
ii、Audio.Settings ………………..………………………………….…………...28
iii、Dial.Settings ……………………………………………………….………...31
iv、Other Settings ………………..………………………….…………….……….32
v、Fxo Port --->Phone Feature……...…………..…….….……………........……….34
CHAPTER 5 Restore Factory Defaults Setting………..…………………37
CHAPTER 6 Upgrade………………………………………………….……...…..38
CHAPTER 7 Ordinary Quality Checking……….…………………………39
CHAPTER 8 FAQ………………………………..………………….......................39

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welcome
The IP voice gateway is the new product. It follows the standard of the
SIP2.0, and is compatible other SIP products or software.
The IP voice gateway is an all-in-one VoIP integrated access device that
features superb audio quality, rich functions, and high level of integration,
compactness and ultra-affordability.

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CHAPTER 1 Installation
What’s in ATA package?
1)One ATA
2)One universal power adapter
3)One Product Enchiridion
4)One Ethernet cable
5)One phone line
Interconnection diagram of ata

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WARNING:
Please do not try to use other power adapter or we would not bear the responsibility to
the defective products.
Changes or modifications to this product not expressly approved by our company, or
operation of this product in any way other than as detailed by this User Manual, could
void your manufacturer warranty.
CHAPTER 2 Production Summarize
I、Key features
•Support SIP2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP,
ARP/RARP, DHCP, NTP, PPPoE, STUN and TFTP etc;
•Inside Rooter、NAT、gateway and DMZ Port transmit;
•Support the PSTN making or receiving calls (by the FXO Port);
•Use the DSP CNOS chip to keeping a wonderful audio-quality,
advanced shaking control, and the technology of hiding the lost
massage;

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•Support kinds of voice coding, including G.711 (a-law and u-law)
G.723.1 (5.3K/6.3K) G.726 (40K/32K/24K/16K), G.729A/B, and
iLBC;
•Support incoming call on show, restricting, and holding,
disconnection, call transfer, call divert, DTMF, dialing project, etc;
•Support conference call;
•Support passing through and T.38 fax, voice restrain and jerquers,
CNG, echo restrains (G.168), AGC, and DIGEST using MD5 and
MD5-sess;
•Support layer 2 (802.1Q VLAN, 802.1p) and layer 3 (QoS, DiffServ,
ToS);
•Support NAT auto- penetrating, no need to modify the setting of the
NAT;
•Support configuration files by inside IVR equipment, Web browser,
or TFTP and HTTP Center Server;
•Support upgrading encrypts configuration files by TFTP or HTTP;
•Microminiaturize and legerity design (size as a wallet), a voice
gateway that convenient for schlepping
II、Technical parameter
ATA(SIP2.0)
YES
route
YES(V610、V610、V620)
bridge
YES(V610、V620)
DHCP
YES
1WAN/1LAN(V610、V610、
V620)
Ethernet Port
1WAN(V600)
1FXS(V600、V610、V610)
FXS
2FXS(V620)
FXO
1FXO(V610)
Exit
YES(V610、V610)
60×110×25mm(V600)
74×106×25mm(V610)
Size
101 × 101 × 31mm （ V610 、
V620）
316g(V600)
Weight
405g(V610)

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423g(V610、V620)
Active status temperature
0—40℃
Humidity
10％—95％
Power supply adapter
AC IN:100V-240V
DC OUT:+9V/V600mA
Use power：<2.5W
Authentication
FCC/CE
CHAPTER 3 Basic Manipulation
I、Keystrokes and voice prompt
There is a inside voice shortcut menu in ATA, can do look through
fleetly and do simple collocation, press button can enter the voice hint
menu. Press the button or the ‘***’ of the common phone, can enter the
voice shortcut menu.
Menu
Voice Prompt
User’s Options
Dial ‘***’ to Main
Menu
“Enter a Menu Option”
Dial ‘****’ to review the
configuration of the device; or Dial
01 – 06, or 99 menu option; or Dial
‘#’ to return to the Main Menu
Dial ‘01’ to
configure the
DHCP Mode
“Static IP Mode”, or
“Dynamic IP Mode”
Dial ‘9’ to toggle the selection. If
user selects “Static IP Mode”, user
need configure the all IP address
information through menu 02 to 05.
If user selects “Dynamic IP Mode”,
the device will retrieve all IP address
information from DHCP server
automatically when user reboots the
device.
Dial ‘02’ to
configure the IP
address of the
device
“IP Address” + IP
address Such as “IP
Address
xxx.xxx.xxx.xxx”
Enter 12-digit new IP address if in
Static IP Mode. for example the
address is “192.168.1.103”, user
should dial “192168001103”

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Dial ‘03’ to
configure the
subnet mask
“Subnet mask” +
subnet mask
Enter 12-digit new subnet mask
address if in Static IP Mode, for
example the address is
“255.255.255.0”, user should dial
“255255255000”.
Dial ‘04’ to
configure the
default gateway
“Gateway “ + IP
address
Enter 12-digit IP address of the
default gateway if in Static IP Mode.
Dial ‘05’ to
configure the DNS
server
“DNS Server” + IP
address
Enter 12-digit IP address of the DNS
server if in Static IP Mode.
Dial ‘06’ to
configure the
TFTP server
“TFTP Server “ + IP
address
Enter 12-digit IP address of the
TFTP server TFTP server is used to
update the firmware of the device.
Dial ‘47’ to make
a direct IP call
“Direct IP Calling”
When dial ‘47’, user will prompt a
dial tone, then dial the 12-digit IP
address. For Example, user wants
call another SIP phone at
“192.168.1.101”. User should dial
‘47’ and dial ‘192168001101’. (For
detail, see “CHAPTER 3 Make a
Direct IP Call”.)
Dial ‘86’ to check
the voice message
“No Voice Messages”;
or “Voice Messages
Pending”
If there are voice messages, user can
dial ‘9’ and dial pre-configured
phone number to retrieve voice
message.
Dial ‘99’ to reset
the device
“RESET”
Dial ‘9’ to confirm the RESET; or
Enter MAC address to restore
factory default setting (For detail,
see CHAPTER 5)
Dial the invalid
number or keypad
“Invalid Entry”
Automatically return to Main Menu
Note:
Press button can enter the voice shortcut menu. Press it again, can switch
to ‘making IP call’ or tonality adjusting mode, when entered the voice hint

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menu mode.
Button ‘*’ have the same function of the button ‘↓’, can switch to next
menu,
Button ‘#’ can go back to the main menu,
Button ‘9’ is like the button ‘enter’, can notarize the current option.
All the typing digits can be recognized by length, two digits is menu
option, 12 digits is IP address. If key in string digits, the system will
administer the relevant order by its length. If key in the wrong dictate,
cannot be deleted, but the system will hint that you have made a mistake
by voice.
Keyboard input cannot by deleted, but the phone will hint when being
deleted.
II 、Make a call
i、How to make the PSTN call (V610、V610)
Connect it exactly as the scheme. Press *00, change to the PSTN line;
dial the number you want immediately. The PSTN number holds the
line.
ii、How to make a VoIP call
1.Connect it exactly as the scheme. Register to the SIP terrace .Dial the
number you want according to the rules that offered by the SIP Internet
terrace and wait for 4 seconds (default)
2. Connect it exactly as the scheme. Register to the SIP terrace .Dial
the number you want according to the rules that offered by the SIP
Internet terrace and then press ‘#’ (assuming that “Use # as dial key” is
selected in web configuration)
iii、Direct IP call
Connect it exactly as the scheme. Dial the number you want

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according to the rules that offered by the ISP Internet.
Insuring the both sides are in the same network range, according
with one of the following condition. (The IP of ATA in the
following condition means the WAN IP)
The ATA or IP equipment of the both sides must have the public
WAN IP address
The ATA or IP equipment of the both sides must in the same
LAN, and the IP must in the same Internet range. E.g.
192.168.1.10 and 192.168.1.20
Make a call immediately according to the IP address.
Pick up the phone or press the speakerphone
Press ‘***47’
Input the 12 digits IP address of the other side with the port at the end, e.g.
if the IP address is 192.168.1.188, port is 5066, press
192168001188*45066, if the port is 5060, it can be omitted.
Press ‘#’ for affirming sending or wait for 4 seconds for auto-sending
Coding form of the character in most common use
00
0
01
1
02
2
03
3
04
4
05
5
06
6

11
07
7
08
8
09
9
*0
. (Dot)
*1
_（Underline）
*2
- (Hyphen)
*3
@
*4
:(Colon)
21
A
22
B
23
C
31
D
32
E
33
F
41
G
42
H
43
I
51
J
52
K
53
L
61
M
62
N
63
O
71
P
72
Q

12
73
R
74
S
81
T
82
U
83
V
91
W
92
X
93
Y
94
Z
Remember the rules of the coding: A is the first letter of the button ‘2’, so its
coding is ‘21’, B is the second letter of the button ‘2’, so its coding is ‘22’,
and so on.
III、Call features
Keys
Call features
30
Block Caller ID (for all subsequent calls)
*31
Send Caller ID (for all subsequent calls)
*67
Block Caller ID (only once)
*82
Send Caller ID (only once)
*50
Disable Call Waiting (for all subsequent calls)
*51
Enable Call Waiting (for all subsequent calls)
*70
Disable Call Waiting (only once)
*71
Enable Call waiting (only once)
*72
Unconditional Call Forward.
To use this feature, dial ‘*72’and get the dial tone. Then dial the
forward number and ‘#’for a dial tone, then hang up.
*73
Cancel Unconditional Call Forward
To cancel ‘Unconditional Call Forward’, dial ‘*73’and get the
dial tone, then hang up

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*90
Busy Call Forward
To use this feature, dial ‘*90’and get the dial tone. Then dial the
forward number and ‘#’for a dial tone, then hang up.
*91
Cancel Busy Call Forward
To cancel ‘Cancel Busy Call Forward’, dial ‘*91’and get the dial
tone, then hang up.
*92
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Then dial
the forward number and “#” for a dial tone, then hang up.
*93
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then
hang up
Flash/H
ook
When in conversation, this action will switch to the new i incoming
call if there is a call waiting indication.
When in conversation without an incoming call, this action will
switch to a new channel for a new call.
IV、Call function
i、Blind transfer
The function is limited to the VOIP calling with FXS port.Assuming
that A and B are in conversation. A wants to Blind Transfer B to C:
A presses FLASH (on the analog phone, or Hook Flash for old model
phones) to get a dial tone.
Then A dials # then dials C’s number, and then #(or wait for 4
seconds)
A can hang up.
Note: Call Feature has to be set to YES.
A can hold on to the phone and wait for one of the three following behaviors:
A quick confirm tone (temporarily using the call waiting indication tone)
followed by a dial tone. This indicates the transfer is successful (transferee has
received a 200 OK from transfer target). At this point, A can either hang up or
make another call.
A quick busy tone followed by a restored call (on supported platforms only).
This means the transferee has received a 4xx response for the INVITE and we
will try to recover the call. The busy tone is just to indicate to the transferor that
the transfer has failed.
Busy tone keeps playing. This means we have failed to receive the second
NOTIFY from the transferee and decided to time out. Note: this does not

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indicates the transfer has been successful, nor does it indicates the transfer has
failed. When transferee is a client that does not support the second NOTIFY
(such as our own earlier firmware), this will be the case. In bad network
scenarios, this could also happen, although the transfer may has been completed
successfully
ii、Attended transfer
The function is limited to the VOIP calling with FXS port. Assuming
that call CHAPTERY A and B are in conversation. A wants to Attend
Transfer B to C:
A presses FLASH (on the analog phone, or Hook Flash for old
model phones) to get a dial tone
A then dial C’s number then # (or wait for 4 seconds). A and C
now in conversation.
A can hang up.Note: When intended Transfer failed, if A hangs
up, the ATA will ring user A again to remind A that B is still on
the call, by pressing FLASH or Hook again will restore the
conversation between A and B.
iii、Conference call
The function is limited to the VOIP calling with FXS
Port.Assuming that call CHAPTERY A and B are in conversation.
A wants C to join the conversation:
A press ‘FLASH’(HOOK FLASH of common or medieval
phone) to get a dial tone
A then dial #+ C’s number + # (or wait for 4 seconds)，then
talk with C
A press flash, begin a conference call.
iv、VoIP-to-PSTN Calls(V610 FXO port)
To make a VoIP-to-PSTN call, users need to dial the FXO SIP
account phone number first. A ring tone is played once followed by
a dial tone. At this time, users can dial a PSTN telephone number or
a mobile telephone number then # (or wait for 4 seconds). The call
will be established afterwards. If no PSTN number is entered after
the dial tone, V610 will hang up automatically in 10 seconds.
In the web configuration page, if the Route to PSTN field is
configured, the second stage dialing is eliminated. That is, after users

15
dial the FXO SIP account number, the PSTN number will be called
automatically.
v 、PSTN-to-VoIP Calls(V610 FXO port)
To make a PSTN-to-VoIP call, PSTN callers need to originate a call to
the FXO port telephone number first. If no one answers the FXS phone after
4 ring tones, a dial tone is played. At this time, users can dial a VoIP
telephone number then # (or wait for 4 seconds). The call will be established
afterwards. If no VoIP number is entered after the dial tone, V610 will hang
up automatically in 10 seconds.
In the web configuration page, if the Route to VOIP field is configured, the
second stage dialing is eliminated. That is, after users dial the FXO port
telephone number, the VoIP number will be called automatically.
Vi、Route Calls to PSTN(V610 FXO port)
If configured, certain calls will be routed to PSTN line automatically. This
call feature is especially useful for emergency calls or local telephone calls.
To use this feature, users need to specify a prefix or a telephone number in
the Route to PSTN field in the web configuration page. If the dialed digits
match one of the specified prefix, outbound calls will be routed to PSTN
port.
vii、Fax
ATA supports two kinds of fax, T.38 and pass through. According to
different SIP roof, selected audio coding as PCMU or PCMA, then can
realize the fax function.
viii、LED light
Red LED means in order
DHCP fail or WAN cannot connect
Flash every 2 seconds (if DHCP has
set)
ATA fail to register
Flash every 2 seconds (if DHCP has
set)

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Green LED means in order
Hint information waiting
Flash every 2 seconds
Calling
Flash endless
Call stoppage
Flash every second
CHAPTER 4 Configuration Guide
I、Configuring WAN IP by voice prompt
V620 LED explicate
As shown from the above figure ,there are five
indicator light in V620.
1、The first one on the right indicates whether the
gateway has registered to the SIP server. （The
red light flashes if it doesn't register, it doesn't
flash on the contrary. ）
2、The second and the forth on the right indicate
whether the LAN\WAN interface of the gateway
has been connected.(They are connected if the
lights are bright.)
3、The third and the fifth show whether there're
data pass throught the interfaces.
V610 LED explicate
As shown from the above figure ,there are five
indicator light in V610.
1、The first one on the right indicates whether the
gateway has registered to the SIP server. （The red
light flashes if it doesn't register, it doesn't flash on
the contrary. ）
2、The second and the forth on the right indicate
whether the LAN\WAN interface of the gateway
has been connected.(They are connected if the
lights are bright.)
3、The third and the fifth show whether there're data
pass throught the interfaces.
V610 LED explicate
V610 has 4 LED and one the facing of a quilt
1、the facing of a quilt one indicates whether the
gateway has registered to the SIP server. （The
red light flashes if it doesn't register, it doesn't
flash on the contrary. ）
2、The green one indicate whether the
LAN\WAN interface of the gateway
has been connected.( It is connected if the light is
bright.)
3、The left one show whether there're data pass
throught the interfaces.
V600 LED explicate
V600 has 2 LED
1、The left one on the right indicates whether the
gateway has registered to the SIP server. （The red
light flashes if it doesn't register, it doesn't flash on
the contrary. ）
2、The right one show whether there're data pass
throught the interfaces.

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Set the IP collocation of ATA by voice hint
i、DHCP mode
According to the 2.2, enter the menu ‘01’ option by voice hint function,
set to DHCP mode.
ii、Static IP mode
According to the 2.2, enter the menu ‘01’ option by voice hint function,
set to static IP mode, and set the IP address of ATA, subnet mask, and the
IP of gateway in the option of ’02 03 04’.
II、Configuring With Web Browser
ATA series products the inside Web server, it can respond the HTTP
GET or POST request from Web Browser, and it has the inside HTML
page layout, it allow the user setting by the Web Browser of the
Microsoft’s IE and AOL’s Netscape.
The collocation menu of ATA can be visited by WAN or LAN port.
The default gateway IP address of LAN port is：
http://192.168.2.1
Get the IP of WAN port:
Press ‘***’ or press the button of ATA equipment to enter the
voice menu
Press ‘02’ can get the IP in English
Use the WAN port to enter the setting Web page, must change the
option ‘Enable Web Access’ to ‘yes’. Then can enter the Web page
by WAN port.

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The network setting menu of ATA:
Open the IE browser, input WAN IP address, will appear the following
page:
The password must case sensitive, and default is ‘voip’.
Input the correct password in the login page, the Web server of ATA will
appear setting menu.
The following is the definition of the parameter in the setting menu
III、status
MAC Address
The device ID, in HEX format. This is a very important
ID for ISP troubleshooting.
WAN IP Address
This field shows WAN port IP address.
Software Version
Program: This is the main software release; its number
is always used for firmware upgrade.
Registered Status
This field indicates whether the device is registered to
the SIP server.
PPPoE Link Up
This field shows whether the PPPoE connection is
enabled or not.
NAT State
This field shows what kind NAT the HT is connected
to via its WAN port. It is based on STUN protocol.

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Activates Total
Time
This field indicates how long the device has been up
since the last reboot.
IV、BASIC OPTIONS
i、LAN Settings
LAN Subnet Mask
Sets the LAN subnet mask. Default value is
255.255.255.0
LAN DHCP Base IP
Base IP for the LAN port which functions as a
Gateway for the subnet. Default value is
192.168.2.1
DHCP IP Flash Time
Value is set in units of hours. Default value is
120hr (5 Days.) The time IP address are assigned
to the LAN clients
ii、WAN Settings