AT-610 User Manual
02
AT-610 User Manual
VERSION 3.0
2012-05-16
ATCOM TECHNOLOGY CO., LIMITED
AT-610 User Manual
1st、 AT-610’s Network Features.........................................................................................4
1、 The View........................................................................................................................ 4
2、 Interfaces...................................................................................................................... 4
3、 Electricity characteristic.......................................................................................... 4
4、 Software........................................................................................................................ 4
5、 Network......................................................................................................................... 5
6、 Management and Maintenance............................................................................. 5
7、 Protocol.......................................................................................................................... 6
8、 Compliant Standard.................................................................................................. 6
9、 Operating Requirement........................................................................................... 6
10、 Packing List................................................................................................................ 6
11、 Installation................................................................................................................. 7
2nd、 Feet installation instruct.............................................................................................7
3rd、 Keypad of IP Phone.................................................................................................... 10
4th、 Menu Operation........................................................................................................... 12
5th、 Basic functions and operations.............................................................................. 13
1、 Answer the calls....................................................................................................... 13
2、 Make Call.................................................................................................................... 14
3、 Hang up the phone................................................................................................. 14
4、 Call Transfer.............................................................................................................. 14
5、 3-way conference call............................................................................................ 15
6、 Call Hold......................................................................................................................15
7、 Call History................................................................................................................ 15
8、 Call pickup..................................................................................................................16
9、 Join call....................................................................................................................... 16
10、 Redial/Unredial...................................................................................................... 16
11、 vport.......................................................................................................................... 17
12、 Click to dial..............................................................................................................17
13、 Preload Password.................................................................................................. 18
14、 Check the Phone’s IP........................................................................................... 18
15、 How to set the IP address through Keypad................................................ 18
6th、 Web settings................................................................................................................. 18
1、 Current state............................................................................................................. 19
2、 Network....................................................................................................................... 19
3、 VoIP.............................................................................................................................. 22
4、 Advance...................................................................................................................... 25
4.1. SIP........................................................................................................................ 25
4.2. Digital Map........................................................................................................ 28
4.3. Stun..................................................................................................................... 29
4.4. Call Service....................................................................................................... 30
4.5. Audio Settings..................................................................................................31
5、 Dial Peer......................................................................................................................32
6、 Config Manage.......................................................................................................... 35
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7、 Update......................................................................................................................... 35
7.1. Web Update...................................................................................................... 35
7.2. FTP/TFTP Update............................................................................................ 36
7.3. Auto Provisioning............................................................................................ 37
8、 System Manage........................................................................................................38
8.1.
8.2.
8.3.
8.4.
8.5.
8.6.
8.7.
Account Manage.............................................................................................. 38
Phone Book....................................................................................................... 39
Syslog Config................................................................................................... 39
Time Set............................................................................................................. 41
Call Log............................................................................................................... 42
Logout................................................................................................................. 42
Reboot.................................................................................................................42
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1st
1st、
AT-610
AT-610’’s Network Features
1、The View
2、Interfaces
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Power: Output Power: 12VDC, 500mA.
WAN: RJ45 port.
LAN: RJ45 port.
3、Electricity characteristic
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Specialty of electric: output 12V 500mA DC
The network connects: 2 RJ45 connect, a WAN, a LAN
Headset jack : RJ9 jack * 2
4、Software
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Sip 2.0 (RFC3261)
Two lines SIP or support IAX2
STUN
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�
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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).
Support inbound audio, RFC2833 and SIP info , DTMF transmission way
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SIP Call Forward、Call transfer、Call hold、Call waiting, 3-way talking、
Pickup、Join call、Redial、Unredial、Call Park、Vport、Click to dial
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Dial without register
Support Hotline、DND(Do Not Disturb)、Blacklists、Call Limitation、DND、
Incoming list
Dial-peer calling rule, IP to IP call
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SIP server conference
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Phone book with 500 records, 100 answered call、missed call for each
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Support HTTP、FTP TFTP updating the configuration and firmware
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Syslog
Answering machine
Support SNTP client
Telnet, WEB visit terminal
Support different level user management
5、Network
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WAN/LAN:Support bridge or route mode
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Support base of NAT and NAPT
Support PPPoE,(ADSL,cable modem use for internet connecting)
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WAN support Primary and Alter function
WAN support DHCP Client
Qos support Diffserv(option)
Support Network command tool: include ping, trace route, telnet
6、Management and Maintenance
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Support safe mode and firmware updating under safe mode
Support different level user management
Configuration via web , keyboard and command
Firmware and configuration updating via HTTP , FTP and TFTP
Support system log and calling record
configuration auto provision
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7、Protocol
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IEEE 802.3 /802.3 u 10 Base T / 100Base TX
PPPoE:PPP over Ethernet
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SIP RFC3261, RFC 2543
TCP/IP: Transfer Control Protocol/Internet Protocol
RTP: Real-time Transport Protocol
RTCP:RTP Control Protocol
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VAD/CNG
Telnet:remote host access protocol
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DNS: Domain Name Server
TFTP:Trivial File Transfer Protocol
HTTP:Hypertext Transfer Protocol
FTP:File Transfer Protocol
8、Compliant Standard
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CE:EN55024,EN55022
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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
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10
10、
�
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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
Packing List
AT-610 IP phone
Power adaptor (output 12v ,500mA)
Manual CD
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11
11、
Installation
Use Ethernet cable to connect AT-610’s LAN port and your computer. Set
computer’s IP to the network 192.168.1.x or using dynamic obtain IP. Press
"Info" button on AT610 keypad several times,user can see the Gateway,Phone
number(NULL) and IP address. Open web browser and key in the ip address,
user will see the logon page of AT-610, the default username and password is
admin/admin for administrator and guest/guest for guest.
Set up page for VoIP use
userr only:
2nd
2nd、
ion
Feet installation instruct
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 position
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:
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B、Push the other plate into the slot in accordance with the direction of the arrow.
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
Sketch Map of Phone
Description:
Number
1
Function description
Network status indicator
A、 when the network is OK , it’s on
B、when the network can not connect , it blinks
3、5、7
Network mode indicator
A、Number 3 stands for DHCP , when the IP phone get IP address in DHCP Mode .
the indicator is on , or else it blinks
B、Number 5 stands for Static , when the IP phone get IP address in Static mode .
the indicator is on , or else it blinks
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C、Number 7 stands for PPPoE, when the IP phone get IP address in PPPoE mode .
the indicator is on , or else it blinks
2、4、6
Answering mode indicator ,
A、Number 2 stands for answering call with handset
B、Number 4 stands for answering call in hand-free mode
C、Number 6 stands for answering call with headset
8
Mute indicator , when the phone is muted , this indicator is on , when the muted
mode is canceled the indicator will be off
9、11
Protocol indicator
A、Number 9 stands for SIP
� When you enable register and successfully register to SIP server , this
indicator is on
� when you enable register but do not successfully register to server , this
indicator blinks
� when you disable register , this indicator is off
B、Number 11 stands for IAX protocol
� When you enable register and successfully register to IAX server , this
indicator is on
� when you enable register but do not successfully register to server , this
indicator blinks
� when you disable register , this indicator is off
10
Missed call indicator , when there is the missed call , this indicator will be on .
After you checked all the missed calls , this indicator will turn off
12
Voicemail indicator , when there is new voice mail , this indicator will be on , after
you checked all the voicemail , this indicator will turn off
13
incoming call indicator
A、 when the phone is in standby without incoming call , the indicator is off
B、when the Ethernet connection is off , the indicator is on
C、when there is the incoming call , indicator blinks
D、 when there is a new voicemail , the indicator blinks
14
Monday to Sunday indicator , the correct day will be on while others will be off
15
This is the multi function key
A、 In standby mode , press Menu/Enter to enter the Menu
B、In functional menu , press Menu/Enter to confirm selection.
16
Press info to check network status
17
press Pbook to enter the phone book and edit or delete.
18
Press History button to check call history including incoming calls , outgoing calls
and missed calls.
19
Exit button
A、 when you’re dialing , press exit to delete the dialed number and return to and
you can dial a new number
B、in functional menu , press EXIT to exit current menu and return to the upper
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menu
20
conference key , when you want to realize conference call , press CONF key and
key in the third party number
21
Transfer key
22
Press Hold to hold the call when you are talking
23
Press mute , you can hear the voice from the other end but he can not hear you
speaking.
24
Voice volume key
A、 press + to turn up the voice volume
B、press - to turn up the voice volume
25
Numeral leys
26
Hand-free button to enter hand-free mode
27
Dial button , when you finish dialing the telephone number , press this button to
send out the phone number
28
Del button is used to delete a single number or letter
29
Headset button is used to enter headset answering mode
4th
4th、
Menu Operation
Press Menu/Enter to enter menu,default password is 123. Use "delect" button to
delect current value then input new value.
Head Menu
Secondary menu
Level 3 Menu
Level 4 Menu
NETWORK
WAN
STATIC NET
IP
DNS2
DNS
GATEWAY
NETMASK
PPPoE
USER NAME
PASSWORD
SIP
SIP SERVER
REGIST
1 ON
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0 OFF
NUMBER
PASSWORD
ACCOUNT
SYSTEM
TIME SET
SET DEFAULT
REBOOT
DSP
INPUT VOLUME
RING VOLUME
RING TYPE
OUTPUT VOLUME
5th
5th、
Basic functions and operations
1、Answer the calls
When there is an incoming call, AT610 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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phone, juts pick up the handset without press any buttons.
2、Make Call
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、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、Call Transfer
Enable Call Transfer
� Blind transfer:
If A is using AT610 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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� Attended transfer:
Only SIP support attended Transfer. If A is using AT610 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-way conference call
Enable Three Way Call
If A is using AT610 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 Hold
User can hold the current call by pressing “Hold”. And press soft “Hold”
again to resume the call.
7、Call History
AT-610 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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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、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.
9、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.
10
10、
Unredial
Redial/
edial/U
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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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
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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12
12、
Click to dial
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
13、
Preload 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
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
14、
Check the Phone
Phone’’s IP
Press "Info" several times to check the IP.
15
15、
How to set the IP address through Keypad
Regarding AT610 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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6th
6th、
Web settings
Enter AT-610 IP addresses in the web browser to go to the log on page, and key
in the username and password to access AT-610 setting page.
Default username and password is:
Administrator:
Username: admin
Password: admin
Guest
:
guest
Guest:
Username:
Password: guest
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
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
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�
�
�
DNS Domain: input DNS domain name if it’s provided
Primary DNS: Primary DNS address
Alter DNS: Alternative DNS address
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, keep
"ANY" as default
Username: ADSL account user name
Password: ADSL account password
Notice:
1)After configuration setting please click “Apply” to effect the change.
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)LAN in AT610 only support bridege mode not router mode.If you connected
your computer to LAN port of AT610, the computer will obtain an ip in the same
segment with WAN port.
4) 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.
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3、VoIP
�
Register Status: SIP server registration status, if succeed display
Registered,or else display Unregistered.
�
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.
�
�
�
Password: SIP account password.
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.
�
�
�
�
�
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�
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.
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
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Advanced SIP setting
�
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Register Expire Time: register expire time, default is 600 seconds. AT-610
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.
Local Port: Local SIP signal port,default as 5060.
Hotline Number: Set hot line number of each line.
Conference Number: config certain Conference call number.
Enable Keep Authentication: Enable/Disable Keep Authentication System
will take the last authentication field which is passed the authentication by
server to 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
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�
�
�
�
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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
�
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address in via field.
Rtp Encode: enable voice data encryption.
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.
Different VoIP Service providers may provide different modes.
DTMF Mode: Select DTMF sending mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Transport Protocol: Set transport protocols, TCP or UDP.
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.
Enable Conference Num: conference ID.
Subscribe Expire Time: Overtime of resending subscribe packet. Suggest
using the default config.
Signal Encode: enable signal encryption.
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
exceeds 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.
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�
Media Key: voice stream encryption Key.
4、Advance
4.1.
SIP
�
Register Status: SIP server registration status, if succeed display
Registered,or else display Unregistered.
�
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.
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.
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Advanced SIP setting
�
�
�
�
�
�
�
�
Register Expire Time: register expire time, default is 600 seconds. AT-610
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
Local Port: Local SIP signal port,default as 5060.
Subscribe Expire Time: Overtime of resending subscribe packet. Suggest
using the default config.
Conference Number: config certain Conference call number
Enable Keep Authentication: Enable/Disable Keep Authentication System
will take the last authentication field which is passed the authentication by
server to 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.
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� 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.
� Rtp Encode: enable voice data encryption.
�
�
�
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.
Different VoIP Service providers may provide different modes.
DTMF Mode: Select DTMF sending mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Transport Protocol: Set transport protocols, TCP or UDP.
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.
Enable Conference Num: Enable or disable conference number.
Enable Subscribe: Enable Subscribe: Overtime of resending subscribe
packet.
Suggest using the default config.
Signal Encode: enable signal encryption.
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.
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�
Media Key: voice stream encryption Key.
4.2.
Digital Map
Digit map is a set of rules to determine when the user has finished dialing.
AT610 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]
represents any one digit between 0~9
� x
represents the last digit timeout. n represents the time from 0~9
� “Tn”
second, it is necessary. “Tn” must be the last two digits in the entry. If “Tn”
is not included in the entry, we use T0 as default; it means system will send
the number immediately if the number matches the entry.
Example:
[1-8] xxx all number from 1000 to 89999 will be sent immediately.
�
8 digits numbers begin with 9 will be sent immediately.
� 9xxxxxxx
Number 911 will be sent will be immediately.
� 911
3 digits numbers begin with 88with be sent after 4s.
� 88xT4
6611x.T4 holds four seconds send out if the number begins 6611 and
�
five digits.
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Notice
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
4.3.
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
Notice: 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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4.4.
�
�
�
�
�
�
Call Service
No Answer Time: no answer call forward time setting.
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.
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.
� Auto Answer: The phone will auto answer when there is an incoming call.
�
�
�
�
�
�
Accept Any Call: The phone will accept the call even if the called number is
not belong to the phone.
Enable Three Way Call: 3 way conference call.
Mute Mode: The phone will not ring and just the ringer led flashes when
there is an incoming call.
Missed call Led: if choosing yes, the led will not flash when there is a missed
call or missed calls. It only shows missed calls on LCD.
Auto handdown: if choosing yes, the phone will return to standby mode
automatically when call party hands up first in headset talking mode.
Black List: Set Add/Delete Black list, incoming call in these phone numbers
will be refused.
It support below rules,
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� 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 dial 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 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
Note: 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.
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4.5.
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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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.
AT610 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:
� 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
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�
�
�
�
�
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 86755-83018047, 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:
� 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.
� 3T rule: If the call starts with 3, the first 3 will be deleted, and the rest
number with be sent to public SIP2 server.
� 2T rule: if the call starts with 2, the first 2 will be deleted, and the rest
number with be sent to IAX2 Server.
� 123 rule: Dial 123 and will send 8675583018049 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 AT610 will send 8675583018049 to your
server.
�
Phone number:There are two types of matching conditions: one is full
matching, the other is prefix matching. In the full matching, you need input
your desired phone number in this blank, and then you need dial the phone
number to realize calling to what the phone number is mapped. In the prefix
matching, you need input your desired prefix number and T; then dial the
prefix and a phone number to realize calling to what your prefix number is
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�
�
�
mapped. The prefix number supports at most 30 digits.
Destination : Set Destination address. This is optional config item. If you
want to set peer to peer call, please input destination IP address or domain
name. If you want to use this dial rule in SIP2 line, you need input 0.0.0.2
in it. If not config, default sip1 as 0.0.0.0.
Port: Set the Signal port, the default is 5060 for SIP.
Alias: Set alias. This is optional config item. If you don’t set Alias, it will
show no alias.
No
tice
Notice
tice:: There are four types of aliases.
1) add: xxx, it means that you need dial xxx in front of phone number, which will
reduce dialing number length.
2) all: xxx, it means that xxx will replace some phone number.
3) del: It means that phone will delete the number with length appointed.
4) Rep: It means that phone will replace the number with length and number
appointed. You can refer to the following examples of different alias application
to know more how to use different aliases and this dial rule.
� Call Mode: Select difference signal protocol, SIP or IAX2.
� Suffix: Set suffix, this is optional config item. It will show no suffix if you
don’t set it.
� Delete Length: Set delete length. This is optional config item. For example:
if the delete length is 3, the phone will delete the first 3 digits then send out
the rest digits. You can refer to examples of different alias application to
know how to set delete length.
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.
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�
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
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.
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�
�
�
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.
7.3.
�
�
�
�
�
�
�
�
Auto Provisioning
Current Version: the system will display the current version number need to
modify the version id need to more than this number on the config file
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
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�
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
8、System Manage
8.1.
Account Manage
Users can add new account or delete and change existing account.
Keyboard Password:config password that you use keyboard to access the
�
�
menu , must be in number.
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】or 【Delete】. General account can only modify or delete general
account.
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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: ring tone.
If you want to make change on existing account , select the account an click
Modify
Delete
【Modify
Modify】or 【Delete
Delete】. General account can only modify or delete general
account.
Notice
Notice:Maximum records of phone book is 500pcs.
8.3.
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
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levels in debug information: Level 0---emergency: This is highest default debug
info level. You system cannot 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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8.4.
�
�
�
�
�
�
Time Set
Server: type the IP address of time server.
Time zone: select correct time zone in list box.
Timeout: longest response time for SNTP.
Daylight Time set: daylight setting through manual.
Manual Time set: Time setting through manual.
Enable Daylight: Daylight saving time.
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8.5.
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.
8.6. Logout
Log out the configuration mode. If you want to re-configuration the phone,
need to input the user and password to login again.
8.7. Reboot
Reboot IP phone, some setting needs to reboot to make it works. Please
always save config before reboot, otherwise the setting will return to previous
setting.
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