ATA V600, V610, V620, V611 User Manual

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User

Manual

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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

P r o m p t … … … … … … … … … … . . … … … . . … … . … … … … … 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 et t i ng s … …… … ……… …… …… .… …… …… …… …… …. 18 ii 、 Wa n Set ti n gs … ……… ……… …… ….… .… ……… ……… ……… . 19 iii、Nat Settings……………………………….….………………………………21 iv、Other Settings……..……………………………………………………………22 v、 Call Settings(V611Fxo port)………..……..……………………………………22

V、Super Options……..…….…………………………………………………..24

i、 SI P S ett in g s… …… …… …… ……… …… ……… …… ….… ……… .2 4 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 ×11 0×2 5mm (V6 00) 74 ×10 6×2 5mm (V6 10)

Size

10 1 × 1 01 × 31 mm （ V6 10 、 V6 20）

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.

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

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. (Dot)

_（Underline）

- (Hyphen)

:(Colon)

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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

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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

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

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PPPoE

The accounts and password of PPPoE.

IP Address

There are 2 modes under which the V600 ata can

operate:

If DHCP mode is enabled then all the field values for

the static ip mode are not used,The V600 will acquire its

IP address from the first DHCP server it discovers from

the connected.

To user the PPPOE feature the PPPOE account

settings need to be set, the V600 will attempt to establish

a PPPOE session if any of the PPPOE field is set

If static IP mode is enabled then IP address 、Subnet

Mask、 Default Router IP address、 DNS Server1、DNS

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Server2 field will need to be configured .These field are

reset to zero by default.

Cloned WAN

MAC Addr

Allow the user to set a specific MAC address. Set in Hex

format

Web server

port

Default for HTTP is 80

WAN ICMP

safety

Unit will not respond to PING from WAN side if set to

“Yes”

iii、NAT Settings

Device Mode

Default to Router mode ,or set up to bridge mode as the

connection of two Ethernet ports.

DMZ IP

Forward all WAN IP traffic to a specific IP address if no

matching port is used by V600 itself or in the defined

port forwarding

Port Map

Allow the user to forward a matching(TCP/UDP) port to

a specific LAN IP address with a specific(TCP/UDP)

port

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Iv、Other Settings

PSTN Access Code

Key pattern to use PSTN line, default is "*00"

Time Zone

This parameter controls how the displayed

date/time will be adjusted according to the

specified time zone.

Daylight Saving Time

This parameter controls whether the displayed

time will be daylight savings time or not. If set

to Yes, then the displayed time will be 1 hour

ahead of normal time.

Optional Rule

Default is 4.1.7.2.0:10.-1.7.2.0:60

Basic User Password

This contains the password to access the Basic

Options. This field is case sensitive.

V、Call settings(V610 FXO PORT)

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The FXO allows to

connect immediately

Choosing yes, The phone switch to voip mode in

on time while change now. while choosing no,

Number of rings before PSTN incoming call is

forwarded. Default is 4.

PSTN Access Key

This field allows users to customize their own

code to access the PSTN line. Default is “*00”.

VOIP CALL PSTN

KEY

Key pattem to authorize calling PSTN numbers

from VOIP,no default

PSTN CALL VOIP

KEY

Key pattem to authorize calling VOIP numbers

from PSTN,no default

Route Call to PSTN

If the dialed digits match one of the specified

prefix here, outbound calls will be routed to PSTN

port. This field is especially useful for emergency

calls.

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Forward to PSTN

Calls are unconditionally forwarded to the

specified PSTN phone number once users dial the

FXO port VoIP number.

Forward to VoIP

Calls are unconditionally forwarded to the

specified VoIP phone number once users dial the

FXO port PSTN number.

V、SUPER OPTIONS

i、SIP Settings

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SIP Server Address

SIP Server’s URI or IP address

Outbound Proxy

SIP Outbound Proxy Server’s URI or IP address

SIP User ID

SIP service subscriber’s User ID

Account ID

SIP service subscriber’s Authenticate ID. Can be

identical to or different from SIP User ID

Authentication Password

SIP service subscriber’s account password

Display Name

SIP service subscriber's name which will be used

for Caller ID display

Home NPA

Default is Black.

Use DNS SRV

Default is No. If set to Yes the client will use

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DNS SRV for server lookup

User ID is phone number

If the ATA has an assigned PSTN telephone

number, this field should be set to “Yes”.

Otherwise, set it to “No”. If “Yes” is set, a

“user=phone” parameter will be attached to the

“From” header in SIP request

SIP Registration

This parameter controls whether the ATA needs

to send REGISTER messages to the proxy server.

The default setting is “Yes”

Unregister On Reboot

Default is No. If set to yes, the SIP user will be

unregistered on reboot

SIP INFO Safety

If set to"YES",the device will ignore any SIP

message that does not come from the IP address

(Source IP in the IP header) that it is registered

to.Default setting is "NO"

This parameter allows the user to specify the time

frequency (in minutes) the ATA refreshes its

registration with the specified registrar. The

default interval is 3V600seconds. The maximum

interval is about 45 days

Local SIP Port

This parameter defines the local SIP port the

ATA will listen and transmit. The default value is

5060(V611FXO port and V620FXS2 port default

5062)

Local RTP Port

This parameter defines the local RTP-RTCP port

pair the ATA will listen and transmit. The default

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value is 5004(V611FXO port and V620FXS2

port default 5008)

Use Random Port

This parameter, when set to Yes, will force

random generation of both the local SIP and RTP

ports. This is usually necessary when multiple

ATA’s are behind the same NAT

NAT Traversal

This parameter defines whether the ATA NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, and then the ATA will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the ATA will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a PortRestricted Cone, the ATA will attempt to use its mapped public IP address and port in all its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the ATA will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open Keep Connected Interval

This parameter specifies how often the ATA sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open Use NAT IP

NAT IP address used in SIP/SDP message. Default is blank Proxy-Require

SIP Extension to notify SIP server that the unit is

behind the NAT/Firewall

DTMF Send Type

This parameter controls how DTMF events are

transmitted. There are 3 ways: in audio which

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means DTMF is combined in audio signal (not

very reliable with low-bit-rate codec), via RTP

(RFC2833), or via SIP INFO

DTMF Payload Type

This parameter sets the payload type for DTMF

using RFC2833

Caller ID Scheme

Bellcore (North America)

ETSI-FSK (France, Germany, Norway, Taiwan,

UK-CCA)

ETSI-DTMF (Finland, Sweden

Denmark-DTMF

CID-Canada

DTMF-Sweden

DTMF-Brazil

Send Anonymous

If this parameter is set to “Yes”, the “From”

header in outgoing INVITE message will be set

to anonymous, essentially blocking the Caller ID

from displaying

Send Flash Event

This parameter allows users to control whether to

send an SIP NOTIFY message indicating the

Flash event, or just to switch to the voice channel

when users press the Flash key

Fax Mode

T.38 mode and pass-through mode

ii、Audio Settings

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Preferred Codecs

The ATA supports up to 7 different Codecs types

including G.711A-/U-law，G.723.1, G.726, G.728,

G.729A/B, iLBC. Depending on the product model,

some of these Codecs may not be provided in

standard release.

Users can configure Codecs in a preference list that

will be included with the same preference order in

SDP message. The first Codecs in this list can be

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entered by choosing the appropriate option in

‘Choice 1’. Similarly, the last Codecs in this list can

be entered by choosing the appropriate option in

‘Choice 7’

G723 Rate

This defines the encoding rate for G723 vocoder. By

default, 6.3kbps rate is chosen

iLBC Frame Size

This sets the iLBC size in 20ms or 30ms

iLBC Payload Type

This defines payload time for iLBC. Default value is

98. The valid range is between 96 and 127

Layer 3 QoS

This field defines the layer 3 QoS parameter which

can be the value used for IP Precedence or Diff-Serv

or MPLS. Default value is 48

Layer 2 QoS(VOIP)

This contains the value used for layer 2 VLAN tag.

Default setting is blank

Layer 2 QoS(PC)

This contains the value used for layer 2 VLAN tag.

Default setting is blank

Silence Suppression

This controls the silence suppression/VAD feature

of G.723 and G729. If set to “Yes”,

when a silence is detected, small quantity of VAD

packets (instead of audio packets)

will be sent during the period of no talking. If set to

“No”, this feature is disabled

Sending or

Reveiving Volume

It can be adjusted the volume in the process

Call progress Tones

It fits to the signal of different countries and areas.

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iii、Dial Settings

Dial Prefix

Sets the prefix added to each dialed number

Dial Plan

Sets the dial plan add to dialed number

Offhook Auto-Dial

This parameter allows a user to configure a User

ID or extension number to be automatically

dialed upon offhook. Please note that only the

user CHAPTER of a SIP address needs to be

entered here. The ATA will automatically

append the “@” and the host portion of the

corresponding SIP address

Use # as Dial Key

This parameter allows the user to configure the

“#” key to be used as the “Send”(or “Dial”) key.

Once set to “Yes”, pressing this key will

immediately trigger the sending of dialed string

collected so far. In this case, this key is

essentially equivalent to the “(Re)Dial” key. If

set to “No”, this # key will then be included as

CHAPTER of the dial string to be sent out

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Enable Call Feature

Default is No. If set to Yes, Call Forwarding &

Do-Not-Disturb are supported locally

Disable Call-Waiting

Default is No

Use Bell-style 3-way

Conference

If "Yes",# will be disabled

iv、 Other Settings

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Firmware Upgrade

This configures IP address of TFTP server. For

upgrading software, ATA can set a TFTP server

of updating code, use LAN or WAN IP address,

and ATA can work in static IP or DHCP mode.

Suggestion: TFTP server must WAN IP address

or in the same LAN with the ATA

NTP server

The parameter defines URL or IP address of NTP

server, ATA show the current date and time

Syslog Server

The IP address or URL of system log server.This

feature is especially useful for ITSP(Internet

Telephone Service Provider)

Syslog level

Select the ATA to report the log level.Default is

NONE.The level is one of

DEBUG ,INFO,WARNING or ERROR.Syslog

message are sent based on the following events:

1. Product model/Version on boot up(INFO

level)

2. Sent or received SIP message(DEBUG level)

3. SLIC chip exception(Warning and error

levels)

4. Memory ecxception(ERROR level)

Subscribe for MWI

Default is NO. When set to Yes a SUBSCRIBE

for Message Waiting Indication will be sent

periodically

FXS Impedance

Selects the impedance of the analog telephone

connected to the Phone port

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Special Feature

Default is Standard.Choose the selcetion to meet

some special requirements from Soft Switch

vendors like Nortel,Broadsoft,etc

Onhook Threshold

The amount of time the hookflash is pressed that

will cause the device to onhook.Default is 800ms

Onhook Voltage

Select the onhook voltage to suit different area or

PBX

Polarity Reversal

Select polarity reversal to calculate the cost for

some system, default is No

Lock Keypad Update

If this parameter is set to "Yes",the configuration

update via keypad is disabled

Super Password

Administrator password.Only administrator can

configure the "Advanced Settings"page.This field

is case sensitive and the maximum password

length is 25 characters.

V FXO PORTPhone Feature

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Early Dial

Default is No. Use only if proxy supports 484

response

Dial Plan Prefix

Sets the prefix added to each dialed number

Use # as Dial Key

This parameter allows the user to configure the

“#” key to be used as the “Send”(or “Dial”) key.

Once set to “Yes”, pressing this key will

immediately trigger the sending of dialed string

collected so far. In this case, this key is

essentially equivalent to the “(Re)Dial” key. If

set to “No”, this # key will then be included as

CHAPTER of the dial string to be sent out

Offhook Auto-Dial

This parameter allows a user to configure a User

ID or extension number to be automatically

dialed upon offhook. Please note that only the

user CHAPTER of a SIP address needs to be

entered here. The ATA will automatically

append the “@” and the host portion of the

corresponding SIP address

SUBSCRIBE for MWI

If set “Yes”, send periodic SUBSCRIBE for

Message Waiting Indication

PSTN Silence Timeout

Terminate call after long silence detected, default

is 60 sec, max 65536

PSTN Disconnect Tone

This configuration should be configured by the

VOIP service provider.Some country use single

frequency tone to signal PSTN

disconnection,some country use double

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frequency tone.

PSTN Disconnect Tone

Cadence

This setting can be configured to suit the

telephone company's standard in different

country.

Once a change is made, the user should press the “Update” button in the

Configuration Menu. The ATA will then display the following screen to

confirm that the changes have been saved

Users are recommended to power cycle the ATA after seeing the above

message

The administrator of the ATA can remotely reboot the ATA by pressing the

“Reboot” button at the bottom of the configuration menu. Once done, the

following screen will be displayed to indicate that rebooting is underway

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At this point, the user can relogin to the ATA after waiting for about 30

seconds.

CHAPTER 5 Restore Factory Default Setting

Warning: Restore the Factory Default Setting will delete all

configuration information of the device.

Step one: Find the MAC Address of the device. The MAC address of the

device is located on the bottom of the device. It is a 12 digits number.

Step two: Encode the MAC address. The encode rule is:

 “2” is the first letter on the button “2” so its encoding is “2”.

 “A” is the second letter on button “2” so its encoding is “22”.

 “B” is the third letter on button “2” and its encoding is “222”.

 “C” is the fourth letter on button “2” and its encoding is

“2222”.

For example, for MAC address 000b8200e395, the user should encode it

as “0002228200333395”.

Step three: Access the voice menu, then dial “99” and get the voice prompt

“RESET”

Step four: Dial in the encoded MAC address. Once the correct encoded

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MAC address is dialed in, the device will reboot automatically and restore

the factory default setting.

Chapter 6 Upgrade

The 6xx Series Products support two types of update modes: TFTP and

HTTP

Attention : Please clear the SIP Server address and Account before remote update; Disconnect to the internet before local update (optional).

The whole process will cost at least 10 minutes, in the period the power

can never be cut off and any manipulation to the phone is not allowed.

to avoid physical break to the gateway , do not cut off the power immediately after the reboot.

一、 TFTP UPDATE

V6xx Series products can work both in DHCP and Static IP mode. The TFTP server is supposed to have a public IP address or the TFTP server and the V6xx products are in the same LAN.

1 LOCAL UPDATE

1) Set up a TFTP server, put the update software package to the root directory of this server, open the TFTP server.

2) Open the configuration page of the gateway, find the place as the following picture:

1. Please specify the Update Mode as :TFTP

2. Fill the Firmware Server item with the Local TFTP server.

3. Click SaveSet then reboot.

二、 HTTP UPDATE

V6xx Series products can work both in DHCP and Static IP mode. The

HTTP server is supposed to have a public IP address or the TFTP server

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and the V6xx products are in the same LAN.

1 LOCAL UPDATE

1) Set up a HTTP server, put the update software package to the root

directory of this server, open the HTTP server.

2) Open the configuration page of the gateway, find the place as the following picture:

1. Please specify the Update Mode as :HTTP

2. Fill the Firmware Server item with the Local HTTP server.

3. Click SaveSet then reboot.

HFS—HTTP Server 2.0a is commended for remote software.

2 REMOTE UPDATE

Please contract us to get a TFTP Server address if you need remote

update.

CHAPTER 7 Ordinary Quality Checking

 Bobbling the ATA, no exceptional noise。

 Use the standard electrical source, connect it without any other

equipment,If the electrical source is OK, you will see: the red LED light will on, after a sound 'pa', it will shine slowly.

CHAPTER 8 FAQ

 、I can open ‘Status’ and ‘Basic setting’, but cannot open the ‘Super options’, why?

 Because when you input the password, you make a mistake. There are

two password, one is super user password (default is ‘admin’), it can open all the pages. The other one is user password (default is ‘123’), it cannot open the ‘Super options’ page.

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 、There is no voice hint after pressing ‘***’ on the phone, why?

 First, checking the parameter on the configuration pages, whether there

is a parameter has been set as ‘*’ or ‘**’ or ‘***’, if have, please change to others’

 Then, ‘Offhook Auto-Dail’ parameter must be empty.

 、The ATA got an IP, but I cannot open the configuration pages, why?

 First, make sure that your computer and the ATA connect the NET.  Then, make sure that your computer and the ATA are in the same

segment.

 Last, make sure that the phone connected to the ATA is onhook.

 、I can visit Internet when I make a VOIP call, why?

 It’s always caused by the bandwidth. Please change the Voice Code, so

that it cost less bandwidth.

 、When I press Speaker or make a call, there is some cacophony, why?

 First, make sure that all the equipments are OK, then, debug the

impedance and onhook voltage of FXS port.

 、The ATA cannot get IP automatically, why?

 First, check the way of connecting; make sure it’s no problem,  Then, visit the configuration through LAN port, make sure all the

equipments in the LAN are use DHCP.

 、The IP address of the ATA is 0.0.0.0, why?

 Main problem: some thing wrong with the connecting, or there is no

DHCP in the LAN.

 、Cannot visit the configuration pages through WAN port, why?

 First, the option on configuration pages called ‘Enable WAN Web

Access’, when you choose ‘no’, you cannot visit the configuration pages through WAN port.

 Then, checking the connection, make sure it is no problem; click Start

then Run, input ‘ping (WAN IP) –l 65000 –t’ order, it can help you to check the status of the NET.

 、Use PPPOE, the ATA can make a VOIP call, but cannot connect the

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NET, why?

 Please set the DNS Server yourself.

 、Cannot visit the configuration pages through LAN port, why?

 Make sure that all the connecting is OK,  Pull out the connection to the WAN port (if the SIP Account is

invalidation, but the connection to the WAN port is OK, it will make the ATA power off automatically.

 Set the computer as DHCP, then reboot the ATA  Check the default gateway, the default gateway of the computer is the

LAN address of ATA