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UTT-562
User Manual
97 pgs5.99 Mb0

Ultrative UTT-562 User Manual

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Shenzhen Ultrative Technology Co., Ltd.
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
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Key
Key name
Function Description
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Navigation
These keys are used in many areas of phone operation.
Depending on the application they will have different
functions. They may be configured through the web page.
directory
Use this key to access the phone book. Records may then
be displayed, edited or deleted. New records may also be
added. To exit Phone Book mode, press and hold this key.
Mute
During a call press this key to prevent the distant party from
hearing the conversation. The distant party will still be
heard.
Line1/2/3/4
The user may select either of them to place a call if they
have been registered to a SIP server.
Volume -/+
Adjust the volume by pressing these two keys.
Redial When off hook, this will dial the last called number.
In stand-by mode, it will check the Outgoing Call.
Speaker
phone
Activate speakerphone mode.
Indicator
light
This light blinks to indicate a missed call.
Soft key 1/2/3/4
Various functions depending on the phone mode.
Description will be shown in LCD.
Call logs
View Missed Calls, Incoming Calls and Outgoing Calls
Keyboard
Dial phone numbers
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DSS keys
Various functions which can be configured in the web interface. See Section 8.3.5.
Port
Port name
Description
Power switch
output: DC 5V AC, 1A WAN
10/100M Connect it to Network
LAN
10/100M Connect it to PC
External console
interface
Port type: RJ-11 direct connector Handset
Port type: RJ-9 connector
Headset
Port type: RJ-9 connector
Icon
Description
Call out
Call in
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Call hold
Auto answer
Call mute
Contact
DND(Do not Disturb)
In hand free mode
In handset mode
In headset mode
SMS
Missed call
Call forward
LED Status
Description
Steady green
The object is idle.
Slow blinking red
The object is ringing.
Steady red
The object is active.
Off
Not subscribed or the object has failed.
LED Status
Description
Steady green
The object is online.
Slow blinking red
The object is ringing.
Steady red
The object is active.
Off
Not subscribed or the object has failed.
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LED Status
Description
Steady green
The account is active.
Fast Blinking Green
There is an incoming call to the account.
Slow Blinking Green
The call is on hold.
Slow Blinking Red
Registration is unsuccessful.
Off
The line is not subscribed or idle.
LED Status
Description
Blinking Green
There are new voice mails.
Off
There is no new voice mail.
LED Status
Description
Steady red
Power on.
Blinking red
There is an incoming call.
Off
Power off.
LED Status
Description
Blinking red
There is an incoming call
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
Shenzhen Ultrative Technology Co., Ltd.
T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
http://www.ultrative.cn | com
Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
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T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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T: +86-755-29685231
E: sales@ultrative.com
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Add: 012#, 5/F, Block A, Hedong Bldg., Liutang Rd., Xixiang St., Bao An District, Shenzhen, China
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Field Name
Explanation
Network
Shows the configuration information for WAN and LAN port,
including connection mode of WAN port (Static, DHCP, PPPoE),
MAC address, IP address of WAN port and LAN port, DHCP server
status for LAN port (ENABLED or DISABLED).
Accounts
Shows the phone numbers and registration status for the 6 SIP LINES
and 1 IAX2 server.
Select the appropriate network mode. The phone supports three network modes:
1 Static: The parameters of a Static IP connection must be provided by your ISP.
2 DHCP: In this mode, network parameter information will be obtained automatically from a
DHCP server.
3 PPPoE: In this mode, you must enter your ADSL account and password.
Refer to Section 2.2 for detailed information about configuring the network parameters.
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8.3.1.2.1 Static IP
If Static IP is selected, this screen will be displayed. Information provided by the ISP should be
entered.
Click Back to return to the Wizard screen. Click Next to go to Quick SIP Settings
8.3.1.2.2 DHCP
After selecting DHCP and clicking NEXT, the Quick SIP Settings screen will appear. Click Back to
return to the Wizard screen. Click Next to go to the Summary screen.
8.3.1.2.3 PPPoE
If PPPoE is selected, this screen will appear. Enter the information provided by the ISP.
Click Back to return to the Wizard screen. Click Next to go to Quick SIP Setting.
8.3.1.2.4 Quick SIP Settings
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Field Name
Explanation
Display Name
The name shown in caller ID.
Server Address
SIP server address either IP address or URI.
Server Port
SIP server port (usually 5060).
Authentication User
Login name or Authentication ID.
Authentication Password
SIP password.
SIP User
Phone number.
Enable Registration
Submits registration information. Normally checked.
Click Back to return to the IP Address screen. Click Next to see summary screen.
Click Finish button to save settings and reboot. After the reboot, SIP calls can be made.
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Field Name
Explanation
Start Time
Start time of the outgoing call
Duration
Duration of the outgoing call.
Dialed Calls
Account, protocol, and line of the outgoing call.
Field name
Explanation
Language
Set the language of phone. English is default.
Greeting Words
The greeting displayed on LCD when phone is idle. It has a
maximum of 12 English characters. Default is VOIP PHONE.
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Field Name
Explanation
Active IP Address
The current IP address of the phone.
Current Subnet Mask
The current Subnet Mask.
Current IP Gateway
The current Gateway IP address.
MAC Address
The MAC address of the phone.
MAC Timestamp
Time the MAC address was obtained.
WAN Settings
The phone supports three network modes. These are also discussed in Section 2.2.
• Static: Network parameters must be entered manually and will not change. All
parameters are provided by the ISP.
• DHCP: Network parameters are provided automatically by a DHCP server.
• PPPoE: Account and Password must be input manually. These are provided by your
ISP.
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Service Name
IP Address or name of DSL Server
User
DSL User Name or Login ID
Password
DSL Password
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Field Name
Explanation
IP Address
LAN static IP.
Subnet Mask
LAN Subnet Mask.
DHCP Service
Activate DHCP server for LAN port. The phone must be rebooted
for the DHCP server setting to take effect.
NAT
Enable NAT operation
Port Mirror
Port Mirror can only be activated in bridge mode. If activated, the
data stream from the WAN port is copied to the LAN port of the
phone.
Enable Bridge Mode
If Bridge Mode is activated, the phone will not provide an IP address
for the LAN port. Instead, the LAN and WAN will be part of the
same network. If this is activated, clicking Apply, will cause the
phone will reboot.
Note: When LAN IP or bridge mode status is changed, the system will reboot! If bridge
mode is chosen, static LAN configuration will be disabled automatically.
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Field Name
Explanation
Enable LLDP
Enable or Disable Link Layer Discovery Protocol (LLDP)
Packet Interval
The time interval for sending LLDP Packets
Enable Learning Function
Enables the telephone to synchronize its VLAN data with the
Network Switch. The telephone will automatically synchronize
DSCP, 802.1p, and VLAN ID values even if these values differ
from those provided by the LLDP server.
Enable DSCP
Enable or Disable Differentiated Services Code Point (DSCP)
SIP DSCP
Specify the value of the SIP DSCP in decimal
Audio DSCP
Specify the value of the Audio DSCP in decimal
Enable WAN Port VLAN
Enable or Disable WAN Port VLAN
WAN Port VLAN ID
Specify the value of the WAN Port VLAN ID. Range is
0-4095
SIP 802.1P Priority
Specify the value of the voice 802.1p priority. Range is 0-7
Audio 8021P Priority
Specify the value of the signal 8021.p priority. Range is 0-7
LAN Port VLAN Mode
Follow WAN: LAN Port ID is same as WAN ID
Disable: Disable Port VALN
Enable: Specify a VLAN ID for the LAN port which is different
from WAN ID
LAN Port VLAN ID
Used when the VLAN ID is different from WAN ID. Range is
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0-4095
Field Name
Explanation
Web Server Type
Specify Web Server Type – HTTP or HTTPS
HTTP Port
Port for web browser access. Default value is 80. To enhance
security, change this from the default. Setting this port to 0 will
disable HTTP access.
Example: The IP address is 192.168.1.70 and the port value is 8090,
the accessing address is http://192.168.1.70:8090.
HTTPS Port
Port for HTTPS access. Before using https, an https authentication
certification must be downloaded into the phone.
Default value is 443. To enhance security, change this from the
default.
Telnet Port
Port for Telnet access. The default is 23.
RTP Port Range Start
Set the beginning value for RTP Ports. Ports are dynamically
allocated.
RTP Port Quantity
Set the maximum quantity of RTP Ports. The default is 200.
Notes:
1. Any changes made on this page require a reboot to become active.
2. It is suggested that changes to HTTP Port and Telnet ports be values greater than 1024.
Values less than 1024 are reserved.
3. If the HTTP port is set to 0, HTTP service will be disabled.
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Field Name
Explanation
DHCP Client Table
IP-MAC mapping table. If the LAN port of the phone connects to a
device, this table will show its IP and MAC address.
Leased Table Name
Name of the lease table.
Start IP Address
Beginning IP address of the lease table.
End IP Address
Ending IP address of the lease table. A device connected to the
LAN port will get an IP address between Start IP and End IP.
Subnet Mask
Subnet Mask of the lease table.
IP Gateway
Network Gateway of the lease table.
Leased Time
Time IP address assignments will persist. Unit is minutes.
DNS Server Address
IP address of DNS server.
Add
Click this button to add this lease table
DHCP Lease Table
Delete
Enter the table name and click the Delete button to remove a DHCP
lease table.
Enable DNS Relay
Activates DNS Relay in the phone. Default is enabled.
Notes:
1. The size of lease table cannot be larger than the quantity of C network IP address. It is
recommended to use the default lease table without modification
2. If the DHCP lease table is modified, the phone must be rebooted.
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Field Name
Explanation
Simple Network Time Protocol (SNTP) Settings
Enable SNTP
Enable or Disable SNTP
Enable DHCP Time
If this is enabled, phone will synchronize time with DHCP server.
Primary Server
IP address of Primary SNTP Server
Secondary Server
IP address of Secondary SNTP Server
Time Zone
Local Time Zone
Resync Period
Time between resync to SNTP server. Default is 60 seconds.
12 -Hour Clock
If checked, clock is 12 hour mode. If unchecked, 24 hour mode.
Default is 24 hour mode.
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Date Format
Specify the date format. Fourteen different formats are available.
Daylight Saving Time Settings
Enable
Enable daylight saving time.
Offset(minutes)
DST offset. Default is 60 minutes.
Month
Start and end month for DST
Week
Start and end week for DST
Day
Start and end day for DST
Hour
Start and end hour for DST
Minute
Start and end minute for DST
Manual Time Settings
Enter the values for the current year, month, day, hour and minute. All values are required.
Note: Be sure to disable SNTP service before entering manual time and date.
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Field Name
Explanation
Choose the sip line to configured (SIP 1 – SIP 6). Click the dropdown arrow to select the
line.
Status
Shows registration status. Will show “Registered” if registered
or “Unapplied” if not registered.
Server Address
SIP server IP address or URI.
Server Port
SIP server port. Default is 5060.
Authentication User
SIP account name (Login ID).
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Authentication Password
SIP registration password.
SIP User
Phone number assigned by VoIP service provider. Phone will not
register if there is no phone number configured.
Display Name
Set the display name. This name is shown on Caller ID.
Enable Registration
Check to submit registration information.
Domain Realm
SIP Domain if different than the SIP Registrar Server.
Proxy Server Address
SIP proxy server IP address or URI（This is normally the same as
the SIP Registrar Server)
Proxy Server Port
SIP Proxy server port. Normally 5060.
Proxy User
SIP Proxy server account.
Proxy Password
SIP Proxy server password.
Backup Server Address
Backup SIP Server Address or URI (This server will be used if the
primary server is unavailable)
Backup Server Port
Backup SIP Server Port
Server Name
Name of SIP Backup server
Codecs Settings
Click on the desired codec to select it. Then use the Left/Right arrow keys to move to the
Enabled or Disabled List. Use the Up/Down arrow to change the priority of enabled codecs.
Advanced SIP Settings
Forward Type
There are 3 call forwarding modes plus Disabled. Disabled：No call forwarding – Default mode Busy：If the phone is busy, incoming calls will be forwarded. No answer：If there is no answer, incoming calls will be forwarded
after a specified time. Always：All incoming calls will be forwarded.
Forward Number
Number to which calls are to be forwarded.
No Ans. Fwd Wait Time
Used in conjunction with Call Forward No Answer. Wait time in
seconds before call is forwarded.
Transfer Timeout
Time interval between sending “bye” message and hanging up
after the phone transfers a call.
Enable Hotline
Activate Hot Line feature. Automatically call a number by going
off hook.
Hotline Number
Number to be called in Hot Line Mode.
Warm Line Wait Time
Used in Hot Line Mode. Time the phone waits after off hook
before dialing the hot line number.
SIP Encryption
Enable/Disable SIP Encryption.
SIP Encryption Key
SIP Encryption key.
RTP Encryption
Enable/Disable RTP Encryption.
RTP Encryption Key
RTP encryption key
Enable Auto Answer
Activate Auto Answer mode. If activated, phone will
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automatically answer an incoming call.
Auto Answer Timeout
Used in conjunction with Auto Answer. The phone will answer
an incoming call after the Auto Answer Timeout
Enable Session Timer
If enabled, this will refresh the SIP session timer per RFC4028.
Session Timeout
Refresh interval if Session Timer is enabled.
Subscribe For MWI
If enabled, the phone will send Message Waiting Indication
(MWI) Subscribe message to the SIP Server
MWI Number
Specify the number to call to retrieve Voice Messages.
Subscribe Period
Time interval between MWI Subscribe Messages.
Conference Type
Choose Conference Type, either local or network
Conference Number
Number to dial to access network conference server. Not needed
if Local conference mode is chosen
Registration Expires
SIP re-registration time. Default is 3600 seconds. If the server
requests a different time, the phone will change to that value.
Enable Service Code
Enables or disables the services described below. These codes
will be sent to the SIP server to activate or deactivate the service.
DND On Code
Do Not Disturb (DND) – When this hot key is pressed, all calls to
the extension to be rejected by the server. The incoming call
record will not be displayed in the Call History.
DND Off Code
Disable Server DND as described above.
Always CFwd On Code
Always Call Forward On – When this function is enabled, the
server will forward all calls to a designated number. The
incoming call record will not be displayed in the Call History.
Always CFwd Off Code
Disable Server Always CFwd as described above.
Busy CFwd On Code
Busy Call Forward On - When this function is enabled, the server
will forward all calls to a designated number if the telephone is
busy. The call record will not be displayed in Call History.
Busy CFwd Off Code
Disable Server Busy CFwd as described above.
No Ans. CFwd On Code
No Answer Call Forward On - When this function is enabled, the
server will forward all calls to a designated number if there is no
answer within a designated time. The incoming call record will not
be displayed in the Call History.
No Ans. CFwd Off Code
Disable Server No Ans. CFwd as described above.
Ban Anonymous On
Code
Ban Anonymous On – When this function is enabled, the server
will disallow the phone to make anonymous calls.
Ban Anonymous Off
Code
Allow Anonymous Calling function described above. In other
words “Anonymous” will be transmitted for Caller ID.
Keep Alive Type
Specifies the NAT keep alive type. If OPTION is selected, the
phone will send OPTION sip messages to the server every NAT
Keep Alive Period. The server will then respond with 200 OK.
If UDP is selected, the phone will send a UDP message to the
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server every NAT Keep Alive Period.
Keep Alive Interval
Set the NAT Keep Alive Interval. Default is 60 seconds
User Agent
Set SIP User Agent value.
DTMF Type
DTMF sending mode. There are four modes:
! In-band (Relay) ! RFC2833 ! SIP_INFO ! AUTO
Different VoIP Service providers may require different modes.
Local port
SIP port. Default is 5060.
Ring type
Set ring tone. There are 9 standard options and 3 user options.
Enable Rport
Enable/Disable support for NAT traversal via RFC3581 (Rport).
Enable PRACK
Enable or disable SIP PRACK function. Default is OFF. It is
suggested this be used.
Enable Long Contact
Allow more parameters in contact field per RFC 3840
Convert URI
Converts # to %23 when sending URI information.
Dial Without Registered
Allow outgoing calls without registration.
Ban Anonymous Call
Refuse Anonymous Calls
Enable DNS SRV
Enables use of DNS SRV records
Enable Missed Call Log
If enabled, the phone will save missed calls into the call history
record.
BLF List Number
BLF List allows one BLF key to monitor the status of a group.
Multiple BLF lists are supported.
Enable BLF List
Enable/Disable BLF List
Server Type
Configures phone for unique requirements of selected server.
RFC Protocol Edition
Select SIP protocol version RFC3261 or RFC2543. Default is
RFC3261. Used for servers which only support RFC2543.
Transport Protocol
Set transport protocol TCP, UDP or TLS.
Anonymous Call Edition
Set privacy support RFC3323, RFC3325 or none
Keep Authentication
Enable /disable registration with authentication. It will use the
last authentication field which passed authentication by server.
This will decrease the load on the server if enabled.
Ans. With a Single Codec
If enabled phone will respond to incoming calls with only one
codec.
Auto TCP
Force the use of TCP protocol to guarantee usability of transport
for SIP messages above 1500 bytes
Enable Strict Proxy
Enables the use of strict routing. When the phone receives packets from the server，it will use the source IP address, not the
address in via field.
Enable GRUU
Support for Globally Routable User-Agent URI (GRUU)
Enable Displayname
Puts quotation marks around the display-name in SIP messages.
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Quote
For servers that require this.
Enable user=phone
Sets user=phone in SIP messages. For compatibility with servers
that require this.
Click to Talk
Set click to Talk (needs support from server).
Respond 182 when Call
waiting
Enable phone responds 182 instead of 180 in some SIP server
envrionment
Use VPN
Enable SIP use VPN for every line individually, not all of them
Enable DND
Enable DND for SIP line individually
SIP Global Settings
Strict Branch
Enable Strict Branch - The value of the branch must be after
“z9hG4bK” in the VIA field of the INVITE message received, or
the phone will not respond to the INVITE.
Note: This will affect all lines
Enable Group
Enable SIP Group Backup. This will affect all lines
Registration Failure Retry
Time
Registration failure retry time – If registration fails, the phone will
attempt to register again after registration failure retry time.
This will affect all lines
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Field Name
Explanation
Status
Shows registration status. Will show “Registered” if registered
or “Unapplied” if not registered.
Server Address
IAX2 server address.
Server Port
IAX2 server port. Default is 4569.
Account
IAX2 account name for registration
Password
IAX2 registration password.
Phone Number
IAX2 phone number (usually the same as IAX2 account name).
Local Port
IAX2 local port. Default is 4569.
Voice Mail Number
Voice mail number.
Voice Mail Text
Voice mail name.
Echo Test Number
If the IAX2 server supports echo test and the echo test number is
non- numeric, this number can be used to replace the echo test
text. This allows dialing a number to perform an echo voice test.
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This function is provided to test whether communication through
the server.
Echo Test Text
Echo test text
Refresh Time
Expiration time of IAX2 server registration. Allowed values are
between 60 and 3600 seconds.
Enable Registration
Enable/Disable IAX2 registration.
Enable G.729AB
Enable/Disable G.729 codec.
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Field Name
Explanation
STUN NAT Transversal
Shows whether or not STUN NAT Transversal was successful.
Server Address
STUN Server IP address
Server Port
STUN Server Port – Default is 3478.
Binding Period
STUN blinding period – STUN packets are sent at this interval
to keep the NAT mapping active.
SIP Waiting Time
Waiting time for SIP. This will vary depending on the
network.
SIP Line Using STUN
SIP Line Using STUN
Select the Line for use with STUN (SIP 1 - SIP 6)
Use STUN
Enable/Disable STUN on the selected line.
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Multiple accounts using the function of dial peer
9T When configuring the SIP1 server, and register，through the sip 1 number beginning
with 9 can achieve the function of del. Example : when dial 98765,it will dial 9876
through the sip1
8T When configuring the SIP2 server, and register，through the sip2 number beginning with
9 can achieve the function of del. Example : when dial 87654,it will dial 8765 through the
sip2
7T When configuring the SIP3 server, and register，through the sip 3 number beginning
with 9 can achieve the function of del. Example : when dial 76543,it will dial 7654
Field Name
Explanation
Phone number
There are two types of matching: Full Matching or Prefix Matching.
In Full matching, the entire phone number is entered and then
mapped per the Dial Peer rules.
In prefix matching, only part of the number is entered followed by
T. The mapping with then take place whenever these digits are
dialed. Prefix mode supports a maximum of 30 digits.
Destination
Set Destination address. This is optional. For a peer to peer call,
enter the destination IP address or domain name. To use a dial rule
on the SIP2 line, enter 0.0.0.2. For SIP3 enter 0.0.0.3
Port
Set the Signaling port, the default is 5060.
Alias
Set the Alias. This is the text to be added, replaced, or deleted. It is
optional.
Note: There are four types of aliases.
1) Add: xxx – xxx will be dialed before any phone number.
2) All: xxx – xxx will replace the phone number.
3) Del: The characters will be deleted from the phone number.
4) Rep: xxx – xxx will be substituted for the specified characters.
Call Mode
Select either SIP or IAX2 protocol.
Suffix
Characters to be added at the end of the phone number. This is
optional.
Delete Length
Sets the number of characters to be deleted. For example, if this is
set to 3, the phone will delete the first 3 digits of the phone number.
This is optional.
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through the sip3
6T When configuring the IAX2 server, and register，through the IAX2 number beginning
with 9 can achieve the function of del .Example : when dial 65432,it will dial 6543
through the IAX2
Dial Peer Examples
Web Interface
Explanation
Example
Set phone number,
Destination, Alias and Delete
Length.
Phone number is XXXT;
Destination is 255.255.255.255
(0.0.0.2) and Alias is del.
Any phone number that begins
with XXX will be sent via
SIP2 after the first several
digits are deleted depending on
the delete length.
Dial “93333”
The SIP2 server will
receive “3333”
This creates a speed dial
function. Dialing “2”, will
cause the entire alias number
to be sent out.
Dial “2”
The SIP1 server will
receive 33334444
The phone will add the alias to
the end of the dialed number if
the dialed number matches the
template in the Phone Number
box.
Dial “8309“
The SIP1 server will
receive “07558309”
Set Phone Number, Alias and
Delete Length. Phone number
is XXXT and Alias is rep: xxx
If the dialed phone number
starts with the digits in the
Phone Number box, the
matching digits will be
replaced by the alias number.
Dial “0106228”
The SIP1 server will
receive “86106228”
If the dialed phone number
starts with the digits in the
Phone Number box, the phone
will send out the dialed phone
number and add the suffix
number.
Dial “147”
The SIP1 server will
receive “1470011”
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Field Name
Explanation
First Codec
The first codec choice: G.711A/u, G.722, G.723, G.729, G.726
Second Codec
The second codec choice: G.711A/u, G.722, G.723, G.729, G.726,
None
Third Codec
The third codec choice: G.711A/u, G.722, G.723, G.729, G.726,
None
Fourth Codec
The forth codec choice: G.711A/u, G.722, G.723, G.729, G.726,
None
Fifth Codec
The fifth codec choice G.711A/u, G.722, G.723, G.729, G.726,
None
Sixth codec
The sixth codec choice G.711A/u, G.722, G.723, G.729, G.726,
None
Onhook Time
Time the handset must be on hook to disconnect a call. Default is
200ms.
Default Ring Type
Ring Sound – There are 9 standard types and 3 User types
Handset Volume
Handset Microphone volume – 9 levels
Speakerphone Volume
Speaker volume in hands free mode - 9 levels
Headset Ring Volume
Headset ring volume – 9 levels
Headset Volume
Headset volume – 9 levels
Speaker Ring Volume
Speaker Ring Volume - 9 levels
G729 Payload Length
G729 Payload Length – Adjusts from 10 – 60 mSec
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Tone Standard
Select tone plan for the country of operation
G722 Timestamps
Choices are 160/20ms or 320/20ms
G723.1 Bit Rate
Choices are 5.3kb/s or 6.3kb/s
Enable VAD
Enable or disable Voice Activity Detection (VAD). If VAD is
enabled, G729 Payload length cannot be set greater than 20 ms.
DTMF Payload Type
The RTP Payload type that indicates DTMF. Default is 101
Enable MWI Tone
Enable special dialing tone as MWI indication, if PBX supports the
special MWI tone indication.
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Field Name
Explanation
Do Not Disturb
DND might be disabled, phone for all SIP lines, or line for SIP
individually.
Enable Call Transfer
If enabled, Call Transfer is allowed.
Semi-Attended
Transfer
If enabled, Semi-Attended Transfer is allowed.
Enable Auto
Handdown
If enabled in speakerphone mode, the phone will automatically hang
up and return to idle when the distant party terminates the call. In
handset mode, it will play dial tone instead of returning to idle.
Auto Handdown Time
Wait time before the phone performs the Auto Handdown behavior
described above.
Enable Auto Redial
If enabled, the phone will automatically redial a call if a busy tone is
received.
Auto Redial Interval
Wait time between auto redial attempts in seconds.
Auto Redial Times
Maximum number of auto redial attempts.
Auto Headset
Automatically answers call on headset.
Enable Intercom
If enabled, allows intercom calls.
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Enable Intercom Tone
If enabled, plays intercom ring tone to alert to an intercom call.
P2P IP Prefix
Set Prefix for peer to peer IP call. For example: You wish to dial
192.168.1.119. If the P2P IP Prefix is defined as 192.168.1., it is
only necessary to dial #119. The default is “.”. If this box is left
blank, IP dialing is disabled.
Turn Off Power Light
Disables Power Light if selected.
Emergency Call
Number
The phone will dial the emergency call number even if the keyboard
is locked. And multi numbers can be added by “,”, such as 911,999
Enable Password Dial
When a number is entered beginning with the password prefix, the
following N numbers after the password prefix will be displayed as
*. N is the value entered in the Password Length field.
For example: If the password prefix is 3 and the Password Length is
2, then dialing the number 34567 will display 3**67 on the phone.
Password Dial Prefix
Prefix for password dialing as described above.
Password Dial Length
Length for password dialing as described above.
Enable Call History
Allow phone to save missed call/dialed call/incoming call or not.
Enable Default Line
If enabled, you can assigned default SIP line for dialing out, not
SIP1.
Alllow IP Call
Allow IP direct call, or disable IP call for dialing.
Play Talking DTMF
Tone
Allow DTMF voice played during talking, or only send DTMF
without local play.
Ban Outgoing
If enabled, no outgoing calls can be made.
Enable Call Waiting
If enabled, notifies user of a second call during a call. Caller ID of
the new caller will be displayed. Press HOLD button to place
existing call on hold and answer new call. Press HOLD again to
return to first call.
Enable 3-way
Conference
If enabled, allows 3-way conference.
Accept Any Call
If enabled, the phone will accept a call even if the called number
does not belong to the phone.
Enable Call
Completion
This is similar to Auto Redial except that the phone detects the state
of the called number before making a new call attempt.
Enable Pre-Dial
If this feature is enabled, digits dialed on-hook will be transmitted
when the phone goes off-hook.
Enable Silent Mode
If enabled, the phone will not ring to indicate a new call. Instead,
the light below the key pad will blink to indicate a new call.
Hide DTMF
This feature sets how DTMF digits are displayed after a call is in
progress. For example, dialing a PIN code to access banking
information. There are 4 choices.
3. Disabled – All the digits will be shown on the LCD.
4. All – None of the digits will be shown on the LCD. The “*”
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will be shown.
5. Delay – The last digit entered will be shown for a short time and
then replaced by “*.”
6. Last Show – The last digit entered will be shown. Previous
digits are replaced by “*.”
Ring from Headset
If this is enabled and a headset is connected, ring tone will be played
in the headset.
Enable Intercom Mute
If enabled, mutes incoming calls during an intercom call
Enable Intercom Barge
If enabled, the phone will auto-answer an intercom call during an
outside call. If an intercom call is established, a second intercom
call will be rejected.
DND Return Code
Specify SIP Code returned for DND. Default is 480 - Temporarily
Not Available.
Busy Return Code
Specify SIP Code returned for Busy. Default is 486 – Busy Here.
Reject Return Code
Specify SIP Code returned for Rejected call. Default is 603 –
Decline.
Active URI Limit IP
IP address of the server for the Action URL messages described
below.
Push XML Server
IP address for XML server which can send display content to the
phone.
Enable Call Waiting
Tone
Enables audible notification of call waiting.
Enable Multi Line
Enable phone to make calls for 10 lines max, or disable for 2 lines
max.
Enable Auto Switch
Line
Enable phone to select an available SIP line as default automatically.
Play Dialing DTMF
Tone
Enable dialing DTMF tone played, or disable for dialing DTMF tone
played.
Action URL Settings
URL for various actions performed by the phone. These actions
are recorded and sent as xml files to the server. Sample format is
http://InternalServer /FileName.xml
Block Out Settings
Add or Delete Blocked numbers – Enter the prefix of numbers
which should not be dialed by the phone. For example, if 001 is
entered, the phone will not dial any numbers beginning with 001.
X and x are wildcards which match single digits. For example, if
4xxx or 4XXX is entered, the phone will not dial any 4 digit
numbers beginning with 4. It will dial numbers beginning with 4
which are longer or shorter than 4 digits.
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Dial Plan Special Characters
[]!
Specifies a range of digits to match. May be a range, a list of ranges separated by
commas, or a list of digits.
x!
Match any single digit that is dialed.
.!
Match any arbitrary number of digits including none.
Tn!
A time out period before digits are sent of n seconds in length. n is mandatory and can
have a value of 0 to 9 seconds. Tn must be the last 2 characters of a dial plan. If Tn is not specified it is assumed to be T0 by default on all dial plans.
，!
Input the digital in front of the ","，then stop and dial tone ，then continue to dial number
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Field Name
Explanation
Phonebook Tables
Group
Dropdown box to select group
Name
Contact name
Office Number, Mobile
Number, Other Number
Contact phone numbers
Ring Type
Ring type for this contact
Group
Contact group for this contact
Add Contact
Name
Contact name
Office Number, Mobile
Number, Other Number
Contact phone numbers
Line
Select line for associated contact number
Ring Type
Ring type for this contact
Group Setting
Choose the group or groups for this contact and move them to the
Selected list on the right.
Import Contact List
Select File
Click the browse button to select the phonebook file to import.
Then click the update button and the selected file will be added to
the phone. File must be xml, vcf or csv format.
Export Contact File
Export XML
Export contacts to xml file.
Export CSV
Export contacts to csv file.
Export VCF
Export contacts to vcf file.
Group Option
Group
Lists existing groups
Name
Enter name for new group
Ring Type
Ring type for group
Blacklist Settings
Type
Select the blacklist type - number or prefix
Value
Input number or prefix
Line
Select the sip line
Note: The maximum capability of the phonebook is 500 contacts.
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Remote XML Phonebook Setting
Note: “x” and “.” are special characters in the black list. “x” matches any single digit and “.”
matches any number of digits. For example, “4xxx” matches any 4 digit number beginning
with 4. “6.” Matches any digit string beginning with 6.
Note: There is also an allowed number list feature if the user only wants to allow a limited
access to the phone. To use this, precede the number with “-“. For example, -123456, or
-1234xx.
Allowed number lists must end with an entry which is only a “.”
This will forbid incoming calls from any number except 4119.
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Phonebook Name
Phonebook name displayed on the phone.
Server URL
Server url of the remote phonebook.
SIP Line
SIP line for the remote phonebook.
Authentication
Authentication mode for remote phonebook.
User/password
Authentication username and password. LDAP Setting
Display Title
The name displayed on the LCD.
Version
LDAP protocol version; supports 3 as default.
Server Address
LDAP server address.
Server Port
LDAP server port.
Authentication
depending on the server’s authentication mode. There are
NONE,DIGEST-MD5,CRAM-MD5,SIMPLE
Line
Assigned SIP line for the LDAP dialing
Username
User name for LDAP authentication
Password
Password for LDAP authentication
Search Base
Search data directory
Enable Calling Search
Allow LDAP contact search and disply LDAP contact name
during dialing and incoming call.
Telephone
Display LDAP contact’s telephone number
Mobile
Display LDAP contact’s mobile phone number
Other
Display LDAP contact’s ohter number
Display Name
Allow display LDAP contact name or not
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MCAST Setting
Prority
Define the priority of the active call, 1 is the highest priority, 10 is the lowest.
Enable Page Priority
The voice call in progress shall take precedence over all incoming paging calls.
Name
Listened multicast server name
Host:port
Listened multicast server’s multicast IP address and port.
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Screen Configuration
Field Name
Explanation
Contrast
Set screen contrast
Enable Backlight
Enable/disable LCD backlight.
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Backlight Time
Set backlight off time.
Line Key Settings/Function Key Settings
Field Name
Explanation
Key
Key Name
Type
Select the type of function the key is to perform. Choices are:
• None
• BLF List Key – If the SIP server supports this function, the key
can monitor the status of a group of phones.
• DTMF – Send DTMF during a call
• Key Event – Many different functions. See Section 4.9.3 for a
list.
• Line – Seize a programmed line (SIP1/SIP2/IAX2)
• Memory Key – See Section 4.9.1.
• URL – Directly access a remote XML phonebook
Value
Parameters associated with the function. For example: The digits to
be dialed by a key programmed for DTMF.
Line
Line on which the function is to be performed.
Subtype
Used with Key Event and Memory Key. Further specifies the type
of function to perform.
Pickup Number
Used with devices which support RFC 5359 Call Pickup
Programmable Key Settings
Key
Choose key to be programmed
Desktop
Choose function in idle mode
Dialer
Choose function while dialing
Calling
Choose function during a call
Desktop Long Pressed
Choose function when key is held down
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Auto Provision Setting
Field Name
Explanation
Current Config Version
Show the current config file’s version. If the version of
configuration downloaded is higher than this, the configuration will
be upgraded. If the endpoints confirm the configuration by the
Digest method, the configuration will not be upgraded unless it
differs from the current configuration.
Common Config
Version
Show the common config file’s version. If the configuration
downloaded and this configuration are the same, the auto provision
will stop. If the endpoints confirm the configuration by the Digest
method, the configuration will not be upgraded unless it differs from
the current configuration.
CPE Serial Number
Serial number of the phone
User
Username for configuration server. Used for FTP/HTTP/HTTPS.
If this is blank the phone will use anonymous.
Password
Password for configuration server. Used for FTP/HTTP/HTTPS.
Config Encryption Key
Encryption key for the configuration file
Common Config
Encryption Key
Encryption key for common configuration file
Save Autoprovision
Information
Save the Autoprovision username and password in the phone until
the server url changes
DHCP Option Settings
Field Name
Explanation
DHCP Option Setting
The phone supports configuration from Option 43, Option 66, or a
Custom DHCP option. It may also be disabled.
Custom DHCP Option
Custom option number. Must be from 128 to 254.
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Plug and Play Settings
Enable PnP
If this is enabled, the phone will send SIP SUBSCRIBE messages to
a multicast address when it boots up. Any SIP server understanding
that message will reply with a SIP NOTIFY message containing the
Auto Provisioning Server URL where the phones can request their
configuration.
PnP Server
PnP Server Address
PnP Port
PnP Server Port
PnP Transport
PnP Transfer protocol – UDP or TCP
PnP Interval
Interval time for querying PnP server. Default is 1 hour.
Phone Flash Settings
Server Address
Set FTP/TFTP/HTTP server IP address for auto update. The address
can be an IP address or Domain name with subdirectory.
Protocol Type
Specify the Protocol type FTP, TFTP or HTTP.
Config File Name
Specify configuration file name. The phone will use its MAC ID
as the config file name if this is blank.
Update Interval
Specify the update interval time. Default is 1 hour.
Update Mode
1. Disable – no update
2. Update after reboot – update only after reboot.
3. Update at time interval – update at periodic update interval
TR069 Settings
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Enable TR069
Enable/Disable TR069 configuration
ACS Server Type
Select Common or CTC ACS Server Type.
ACS Server URL
ACS Server URL.
ACS User
User name for ACS.
ACS Password
ACS Password.
TR069 Auto Login
Enable/Disable TR069 Auto Login.
"Inform" Sending Period
Time between transmissions of “Inform” Unit is seconds.
Level
Name
Description
0
Emergency
System is unusable. This is the highest debug info level.
1
Alert
Action must be taken immediately.
2
Critical
Critical conditions. System is probably working incorrectly.
3
Error
Error conditions. System may not work correctly.
4
Warning
Warning conditions. System may work correctly but needs
attention.
5
Notice
Normal but significant condition.
6
Informational
Normal daily messages.
7
Debug
Debug messages normally used by system designer. This level
can only be displayed via telnet.
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Syslog Configuration
Field Name
Explanation
Syslog Settings
Server IP
Syslog server IP address.
Server Port
Syslog server port.
MGR Log Level
Set the level of MGR log.
SIP Log Level
Set the level of SIP log.
IAX2 Log Level
Set the level of IAX2 log.
Enable Syslog
Enable or disable syslog.
Web Capture
Start
Capture a packet stream from the phone. This is normally used to
troubleshoot problems.
Stop
Stop capturing the packet stream
Config Setting
Field Name
Explanation
Save Configuration
Save the current phone configuration. Clicking this saves all
configuration changes and makes them effective immediately.
Backup Configuration
Save the phone configuration to a txt or xml file. Please note to
Right click on the choice and then choose “Save Link As.”
Clear Configuration
Logged in as Admin, this will restore factory default and remove all
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configuration information.
Logged in as Guest, this will reset all configuration information
except for VoIP accounts (SIP1-6 and IAX2) and version number.
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Update
Field Name
Explanation
Web Update
Web Update
Browse to the config file, and press Update to load it to the phone.
Various types of files can be loaded here including firmware, ring
tones, local phonebook and config files in either text or xml format.
TFTP/FTP Update
Server Address
FTP/TFTP server address for download/upload. The address can be
IP address or Domain name with subdirectory.
User
FTP server Username for download/upload.
Password
FTP server password for download/upload.
File name
Name of update file or config file. The default name is the MAC of
the phone.
Note: The exported config file can be modified. The config file is made up of modules.
Modules which do not need changes may be deleted. For example, a config file can be
downloaded and all modules removed except the SIP module. After rebooting, only the SIP
settings will be changed.
Type
Action to be executed by the phone.
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1. Application update - download system update file
2. Config file export - Upload config file to FTP/TFTP server. It
can then be named and saved.
3. Config file import - Download the config file from FTP/TFTP
server. The configuration will be effective after the phone is
reset.
4. Phone book export (.vcf, .csv, .xml) - Upload the phonebook file
to FTP/TFTP server. It can then be named and saved.
5. PhoneBook import (.vcf, .csv, .xml) - Download phonebook file
from FTP/TFTP server.
Protocol
Select FTP/TFTP server.
Update Logo File
Select File
URL of the logo file.
Delete Logo File
Select File
Logo file name to be deleted.
Logo File
Logo File
Logo file in use.
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Access Configuration
Field Name
Explanation
LCD Menu Password Settings
Menu Password
Sets the password for entering the setup menu from the phone
keypad. The password must be only digits.
User Settings
This table shows the current user accounts
Add User
User
Set User Account name
User Level
There are two levels. Root user can modify the configuration.
General user can only read the configuration.
Password
Set the password
Confirm
Confirm the password
User Management
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Select the account and click Modify to modify the selected account. Click Delete to delete
the selected account.
A General user can only add another General user.
WEB Filter
The Web filter is used to limit access to the phone. When the web filter is enabled, only the
IP addresses between the start IP and end IP can access the phone.
Field Name
Explanation
Start IP Address
Beginning IP Address for MMI Filter
End IP Address
Ending IP Address for MMI Filter
Add
Add this filter range to the Web Filter Table
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Enable Web Filter
Select to enable MMI Filter.
Apply
Make filter settings effective.
Note: Once a range is added, it can be modified or deleted.
Note: Be sure that the filter range includes the IP address of the configuration computer.
Firewall Configuration
Firewall rules can be used to prevent unauthorized Internet users from accessing private
networks connected to this phone (input rule), or prevent unauthorized devices connected to
this phone from accessing the Internet (output rule). Each rule type supports a maximum of
10 items.
Field Name
Explanation
Enable Input Rules
Enable rules limiting access from the Internet.
Enable Output Rules
Enable rules limiting access to the Internet.
Input/Output
Specify if the current rule is input or output.
Deny/Permit
Specify if the current rule is Deny or Permit.
Protocol
Filter protocol type (TCP/ UDP/ ICMP/ IP)
Port Range
Set the filter Port range
Src Address
Set source address. It can be a single IP address or use * as a wild
card. For example: 192.168.1.14 or *.*.*.14.
Dest Address
Set destination address. It can be a single IP address or use * as a
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wild card. For example: 192.168.1.14 or *.*.*.14.
Src Mask
Set the source address mask. For example: 255.255.255.255 points to
one host while 255.255.255.0 points to a C type network.
Dest Mask
Set the destination address mask. For example: 255.255.255.255
points to one host while 255.255.255.0 points to a C type network.
When a connected device tries to access 192.168.1.118, the phone will deny the request
because of the out_access rule. Access to any other IP address will be allowed.
Click the Delete button to delete the selected rule.
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Application Layer Gateway (ALG) Settings
Field Name
Explanation
IPSec ALG
Enable/Disable IPSec encryption. Default is enabled.
FTP ALG
Allow the ALG to securely pass FTP traffic. Default is enabled.
PPTP ALG
Allow the ALG to securely pass PPTP traffic. Default is enabled.
Network Address Translation (NAT) Table
Shows the NAT TCP and UDP mapping tables
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NAT Table Option
Transfer Type
Select the TCP or UDP protocol.
Inside IP
Set the local IP address of device.
Inside Port
Set the LAN (inside) port for NAT mapping
Outside Port
Set the WAN (outside) port for NAT mapping
Note: After entering settings, click the Add button to add new mapping table data. To delete
an entry, enter its information and then click the Delete button.
Notice: The phone supports 10M/100M adaptive. Under loaded conditions traffic through the
phone NAT may not reach 100M.
Field Name
Explanation
VPN Status
Shows the current VPN IP address.
VPN Mode
Select L2TP. You can choose only one for current state. After you select it, save the
configuration and reboot the phone.
Enable VPN
Enable/Disable VPN.
L2TP
Select Layer 2 Tunneling Protocol
OpenVPN
Select OpenVPN Protocol
Only one protocol may be activated. After the selection is made, the configuration should be
saved and the phone rebooted.
VPN Server Address
Set VPN L2TP Server IP address.
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VPN User
Set User Name access to VPN L2TP Server.
VPN Password
Set Password access to VPN L2TP Server.
Field Name
Explanation
Update Security File
Select Security File
Browse to the security file to be updated. Click the Update button to
update.
Delete Security File
Select Security File
Select the security file to be deleted. Click the Delete button to
Delete.
SIP TLS File
Show SIP TLS authentication certificate.
HTTPS File
Show HTTPS authentication certificate.
OpenVPN Files
Show OpenVPN File authentication certificate file.
Item
Specification
Power Adapter
Input: 100-240V
Output: 5V 1A
Port
WAN
10/100Base- T RJ-45 1 PORT
LAN
10/100Base- T RJ-45 1 PORT
EXT
RJ11 1 PORT
Headset
RJ9 1 PORT
Power Consumption
Idle: 2.5W
Active: 2.8W
LCD Size
128x64
53.5 x 70mm
Operation Temperature
0～40℃
Relative Humidity
10～65%
CPU
Broadcom
SDRAM
16MB
Flash
4MB
Dimension(L x W x H)
290x260x60mm
Weight
1.2kg
Keypad
Character
Keypad
Character
1 @
7 P Q R S p q r s
2 A B C a b c
8 T U V t u v
3 D E F d e f
9 W X Y Z w x y z
4 G H I g h i
*/.
5 J K L j k l
0
6 M N O m n o
#/=
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