Basicfunctionsandoperations. ATCOM AT-610P

AT-610P User Manual

SYSTEM

DSP

NUMBER

PASSWORD

ACCOUNT

TIME SET

SET DEFAULT

REBOOT

INPUT VOLUME

RING VOLUME

RING TYPE

OUTPUT VOLUME

0 OFF

1

Answer

When there is an incoming call, AT6I0P will remind user with ring. 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 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 phoning under hand-free mode and want to change to speaker

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AT-610P User Manual phone, juts pick up the handset without press any buttons.

2

Make C all

A、Use the handset

Pick up the handset, the LCD will show the current lines. 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 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 phone.

b、 Press " Pbook” button.

c、 Press “Menu/Enter” to enter the phone list and use "Vol+" or "Vol-" keys to find the contact person.

d、 When you find the certain contact person, press" Menu/Enter" to show the details.

e、 Press "Edit" to edit the number or press" Dial" to call.

3

H ang

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

Call

Enable Call Transfer

� Blind transfer:

If A is using AT6I0P talking with B, B want to speak to C. A just press "XFER" and dial C’s number,after C rings, A will be hung up.

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AT-610P User Manual

� Attended transfer:

Only SIP support attended Transfer. If A is using AT6I0P talking with B , B want to speak to C. A just press "Hold" and dial C’s number to ask whether he would answer the call from B. If C agrees , A press "Hold" to talk with B then press "XFER" to transfer the call.

5

3

Enable Three Way Call

If A is using AT6I0P talking with B and B want to make conference call with

A and C. A just press "Hold" and dial C’s number. Then press "CONF" to initiate conference call.

6

Call

User can hold the current call by pressing “Hold”. And press soft “Hold” again to resume the call.

7

Call

AT-610P 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 display the Missed call icon and the status LED is 2s on,

500ms off.

2) Press history button, press “Vol+” or “Vol-“ to LCD display “Missed”.

3) Press “Menu/Enter” to display the records of Missed call.

4) Press “Vol+” or “Vol-“browse missed call history.

5) Choose the missed call record, press “Redial” button to call this number.

� Incoming call

1) Press “History” button and press “Vol+” or “Vol-“to LCD display “RECEIVED”.

2) Press “Menu/Enter” to display received call records.

3) Press “Vol+” or “Vol-“browse received call records.

4) Choose the received call record, press “Redial” button to call this number.

If there is no record, the LCD screen display “List is Empty”.

� Out coming call

1) Press “History” button and press “Vol+” or “Vol-“to LCD display

“DIALED”.

2) Press “Menu/Enter” to display out coming call records.

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AT-610P User Manual

3) Press “Vol+” or “Vol-“browse out coming call records.

4) Choose the received call record, press “Redial” button to call this number.

If there is no record, the LCD screen display “List is Empty”.

8

C all

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.

9

J oin

“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.

10

R edial/

Unredial

In order to being efficiently to contact the busy line, A could use Redial to call B the busy line with setting prefix. When B is free A could get through the call as usual. When B is busy, A could hang the phone with checking B’s situation with every 60S by the set of prefix.

IP Phone of User A would ring and prompt picking up handset if B is available.

It would call B automatically once A picking up handset. The call would get through as soon as had set being picked up at B. A could dial the predecessor which set already add number of B to cancel the call before the phone automatic redialing if A is not available suddenly or don’t want to call B

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AT-610P User Manual anymore.

*3* is the predecessor. Then A could make the redial function via dialing *3* + number of B.

*4* is the predecessor. Then A could make the redial function via dialing *4* + number of B.

User could name any predecessor like *3*/*4* if it is compliant with present dial rule.

11 vport

Vport makes more flexible calling application. E.g. 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 (e.g.: 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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AT-610P User Manual

12

Click

When User A accesses web interface and calls User B via clicking one link which is direct to B, IP Phone of User A would ring. Then call B automatically once User A picking up handset.

13

Preload

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 model. While user 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 model after input different user name and password:

� Guest Model

� Username：guest

� Password： guest

� Admin Model：

� Username：admin

� Password： admin

14

Check

Press "Info" several times to check the IP.

15

How

Regarding AT6I0P setting Network mode to Static via keypad as below:

Long press "1" after LCD display “Static mode change…” on this time, IP phone is default IP address as 192.168.1.179.

If you need to change Static IP address, please refer as below:

� press “Menu/enter” button and input password as “ 123”

� press “Menu/enter” button that LCD display Network

� Still press enter button until LCD display current static IP address

� press “delete” button that LCD display "input value"

� Input the IP address as you need and press enter after finished input IP address, IP phone will saving the configuration auto after 30 seconds.

� Long press "1" for Static

� Long press "2" for DHCP

� Long press "3" for PPPoE

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