Cortelco IP Phone C58 User Manual

C58/C58P VoIP Phone User  
Manual  
Cortelco  
1703 Sawyer Road  
Corinth, MS 38834  
USA  
Tel: (662)287-5281  
Fax:(662)287-3889  
Version 1.0  
August 14, 2013  
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Table of Contents  
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1
Introducing C58/C58P VoIP Phone  
1.1  
Thank you  
Thank you for purchasing the C58/C58P Voice Over Internet Protocol (VoIP) telephone.  
The C58/C58P is a fully featured telephone that provides voice communication over the data  
network. This phone has all the features of a traditional telephone and gives access to many  
data service features. The C58 and C58P are identical except that the C58P can be powered  
via Power over Ethernet (PoE). This guide will help you easily use the various features and  
services available on your phone.  
1.2  
Box Contents  
The following items should be packed with your telephone. Please contact your dealer if  
any of them are missing.  
Telephone (Main body) with display and keypad  
Handset  
Handset cord  
Power supply  
Ethernet cable  
1.3  
Keypad  
Key  
Key name Function Description  
These keys are used in many areas of phone operation.  
Depending on the application they will have different  
functions.  
Navigation  
They may be configured through the web page.  
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.  
During a call press this key to prevent the distant party from  
hearing the conversation. The distant party will still be  
Directory  
Mute  
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heard.  
Volume -/+  
Redial  
Adjust the volume by pressing these two keys.  
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.  
Various functions depending on the phone mode.  
Description will be shown in LCD.  
Soft key 1/2/3/4  
History  
View Missed Calls, Incoming Calls and Outgoing Calls  
Keyboard  
Dial phone numbers  
Various functions which can be configured in the web  
interface. See Section 8.3.5.  
DSS keys  
1.4  
Input/Output Ports  
Port  
Port name  
Description  
Power switch  
WAN  
Input: 5V AC, 1A  
10/100M Connect to Network  
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LAN  
Headset  
10/100M Connect to PC  
Port type: RJ-9 connector  
Port type: RJ-9 connector  
Handset  
1.5  
Icon Introduction  
Icon  
Description  
Call out  
Call in  
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  
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1.6  
LED Introduction  
1.6.1  
Programmable key LEDs for BLF  
LED Status  
Description  
Steady green  
Slow blinking red  
Steady red  
The object is idle.  
The object is ringing.  
The object is active.  
The object has failed.  
Not subscribed.  
Fast blinking red  
Off  
1.6.2  
Programmable key LEDs for Presence  
LED Status  
Description  
Steady green  
Slow blinking red  
Steady red  
The object is online.  
The object is ringing.  
The object is active.  
The object has failed.  
Not subscribed.  
Fast blinking red  
Off  
1.6.3  
Programmable key LEDs for MWI  
LED Status  
Description  
Blinking Green  
Off  
There are new voice mails.  
There is no new voice mail.  
1.6.4  
Power Indication LED (Power Light Enabled)  
LED Status  
Steady red  
Blinking red  
Off  
Description  
Power on.  
There is an incoming call.  
Power off.  
1.6.5  
Power Indication LED (Power Light Disabled)  
LED Status  
Description  
Blinking red  
There is an incoming call  
2
Initial Connection and Setting  
2.1  
Connecting the phone  
1. Connect to the network. Use the Ethernet cable in the package to connect the WAN port  
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on the back of your phone to an Ethernet port. The following two figures show  
connection options.  
a. Direct network connectionThis method requires at least one available Ethernet port.  
Connect the WAN port on the back of your phone to the Ethernet port. Since the phone  
has a built-in router, it can be connected directly to the network.  
b. Shared network connectionUse this method if you have a single Ethernet port which is  
already in use. Disconnect the Ethernet cable from the Ethernet port and attach it to the  
WAN port on the back of the phone. Then use the Ethernet cable in the package to  
connect the LAN port on the back of the phone to the other device. The IP Phone now  
shares a network connection.  
2. Connect the handset to the handset jack using the handset cable in the package.  
3. Connect the power supply to the DC port on the back of the phone. Connect the  
power supply to a standard power outlet. Note that the power supply will not be  
needed if your network provides Power over Ethernet (PoE).  
4. The phone’s LCD screen displays “INITIALIZING”. Later, a ready screen displays  
the date, time and current network mode.  
If your LCD screen displays different information from the above, more information may  
need to be entered. Please refer to the next section. If your phone registers into your IP  
telephony Server, it is ready to use. If not, continue to read for more configuration  
information.  
2.2  
Network Settings  
DHCP is supported by default. This allows the phone to receive an IP address and other  
network-related settings (Netmask, IP gateway, DNS server) from the DHCP server. If no  
DHCP server is available, the network connection settings must be changed. Follow the  
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instructions below to change to either PPPoE or static IP.  
2.2.1  
PPPoE Mode  
1. Press the MENU softkey.  
2. Scroll down to 3. Settings.”  
3. Press OK.  
4. Scroll down to 2. Advanced Settings.”  
5. Press OK.  
6. The LCD will display “INPUT PASSWORD”.  
7. Input the password (default value is 123).  
8. Press ENTER.  
9. Scroll down to 2. Network.”  
10. Press OK.  
11. Press OK to select WAN Settings.  
12. Scroll down to 4. PPPoE Settings.”  
13. Press OK.  
14. Use the keypad to enter the User Name.  
15. Press SAVE softkey.  
16. Press DOWN ARROW.  
17. Use the keypad to enter the Password.  
18. Press SAVE softkey.  
19. Press DOWN ARROW.  
20. Use LEFT ARROW or RIGHT ARROW to enable PPPoE.  
21. Press SAVE softkey.  
22. Press BACK softkey to return to the WAN Settings screen.  
23. Press UP ARROW or DOWN ARROW to scroll to 1. Connection Mode.”  
24. Press OK.  
25. Use LEFT ARROW or RIGHT ARROW to select PPPoE.”  
26. Press SAVE softkey.  
27. Press BACK or EXIT 6 times to return to idle screen.  
28. Disconnect and reconnect the power supply so the phone will reboot and apply the new  
settings.  
2.2.2  
Static IP Mode  
1. Press the MENU softkey.  
2. Scroll down to 3. Settings.”  
3. Press OK.  
4. Scroll down to 2. Advanced Settings.”  
5. Press OK.  
6. The LCD will display “INPUT PASSWORD”.  
7. Input the password (default value is 123).  
8. Press ENTER.  
9. Scroll down to 2. Network.”  
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10. Press OK.  
11. Press OK to select WAN Settings.  
12. Scroll down to “2. Static IP Settings.”  
13. Press OK.  
14. Use the keypad to enter the IP Address.  
15. Press SAVE softkey.  
16. Press DOWN ARROW.  
17. Use the keypad to enter the Subnet Mask.  
18. Press SAVE softkey.  
19. Press DOWN ARROW.  
20. Use the keypad to enter the Gateway Address.  
21. Press SAVE softkey.  
22. Press DOWN ARROW.  
23. Use the keypad to enter the DNS 1 Address.  
24. Press SAVE softkey.  
25. Press DOWN ARROW.  
26. Use the keypad to enter the DNS 2 Address if desired.  
27. Press SAVE softkey.  
28. Press BACK softkey.  
29. Press UP ARROW or DOWN ARROW to scroll to 1. Connection Mode.”  
30. Press OK.  
31. Use LEFT ARROW or RIGHT ARROW to select “Static IP.”  
32. Press SAVE softkey.  
33. Press BACK or EXIT 6 times to return to idle screen.  
34. Disconnect and reconnect the power supply so the phone will reboot and apply the new  
settings.  
2.2.3  
DHCP Mode  
1. Press the MENU softkey.  
2. Scroll down to 3. Settings.”  
3. Press OK.  
4. Scroll down to 2. Advanced Settings.”  
5. Press OK.  
6. The LCD will display “INPUT PASSWORD”.  
7. Input the password (default value is 123).  
8. Press ENTER.  
9. Scroll down to 2. Network.”  
10. Press OK.  
11. Press OK to select WAN Settings.  
12. Scroll down to “3. DHCP Settings.”  
13. Press OK.  
14. Use LEFT ARROW or RIGHT ARROW to enable or disable DHCP DNS.  
15. Press SAVE softkey.  
16. Press DOWN ARROW.  
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17. Use LEFT ARROW or RIGHT ARROW to enable or disable DHCP Time.  
18. Press SAVE softkey.  
19. Press BACK softkey.  
20. Press UP ARROW or DOWN ARROW to scroll to 1. Connection Mode.”  
21. Press OK.  
22. Use LEFT ARROW or RIGHT ARROW to select “DHCP.”  
23. Press SAVE softkey.  
24. Press BACK or EXIT 6 times to return to idle screen.  
25. Disconnect and reconnect the power supply so the phone will reboot and apply the new  
settings.  
3
Basic Functions  
Making a call  
Call Device  
3.1  
3.1.1  
Calls can be made using three different devices:  
1. Handset - Pick up the handset. The  
icon will be shown on the LCD screen.  
2. Speakerphone - Press the Speaker button. The  
icon will be shown on the LCD  
screen.  
3. Headset - Press the Headset button. The icon  
screen.  
will be shown in the LCD  
The number may also be dialed first. Then the method of speaking can be chosen.  
3.1.2  
Call Methods  
Press an available line button then use one of the following methods to place a call.  
1. Dial the desired number using the keypad.  
2. Press the History softkey. Use the navigation buttons to highlight the number to call.  
Use the LEFT ARROW or RIGHT ARROW to choose Missed Calls, Incoming Calls and  
Outgoing Calls.  
3. Press the REDIAL button to redial the last number called.  
4. Press a programmable key which has been configured as a speed dial button.  
5. Press the Dial softkey to make the call if necessary.  
3.2  
Answering a call  
If the phone is idle, lift the handset, press the Speaker button or Answer softkey to answer  
using the speaker phone, or press the headset button to answer using the headset.  
If the phone is in use, press the answer softkey.  
During the conversation, you can alternate between headset, Handset and Speaker phone by  
pressing the corresponding buttons or picking up the handset.  
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3.3  
Do Not Disturb (DND)  
Press the DND softkey to active DND Mode. New incoming calls will be rejected and the  
display will show: icon. Press the DND softkey twice to deactivate DND mode.  
Incoming calls will be stored in the Call History.  
3.4  
Call Forward  
This feature allows forwarding an incoming call to another phone number. The display shows  
icon.  
The following call forwarding events can be configured:  
Off: Call forwarding is deactivated by default.  
Always: Incoming calls are immediately forwarded.  
Busy: Incoming calls are immediately forwarded when the phone is busy.  
No Answer: Incoming calls are forwarded when the phone is not answered  
after a specific period.  
To configure Call Forward via Phone interface:  
1. Press Menu ->Features->OK>Call Forwarding->OK.  
2. Select the line to be forwarded.  
3. Use LEFT ARROW or RIGHT ARROW to select Disabled, Always, Busy, or No Answer.  
4. After choosing a mode (except Disabled), press DOWN ARROW and then enter the  
phone number for forwarding.  
5. Press Save to save the changes.  
3.5  
Call Hold  
1. Press the Hold softkey to put the active call on hold.  
2. If there is only one call on hold, press the Hold softkey to retrieve the call.  
3. If there is more than one call on hold, press the line button, and the Up/Down button to  
highlight the call, then press the Resume button to retrieve the call.  
3.6  
Call Waiting  
1. Press Menu ->Features->Enter->Call Waiting->Enter.  
2. Use the navigation keys to activate or deactivate call waiting.  
3. Press SAVE to save the changes.  
3.7  
Call Waiting Tone  
1. Press Menu ->Features->Enter->Call Waiting Tone->Enter.  
2. Use the navigation keys to activate or deactivate call waiting tone.  
3. Press SAVE to save the changes.  
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3.8  
Mute  
When the Mute button is pressed during a conversation, the  
icon will be shown in the  
LCD. The distant party will not hear the party on the C58/C58P, but the distant party can  
still be heard.  
Press Mute again to return to normal conversation.  
3.9  
Call transfer  
3.9.1  
Blind Transfer  
During a conversation, press the XFER key, dial the number to which the call is to be  
transferred followed by "#" and then hang up.  
3.9.2  
Attended Transfer  
During a conversation, press the XFER key, dial the number to which the call is to be  
transferred followed by "#" and press Send. After the third party answers, press XFER to  
complete the transfer.  
NOTE: Call waiting and call transfer must be enabled.  
NOTE: The SIP server must support RFC3515.  
3.9.3  
Semi-Attended Transfer  
During a conversation, press the XFER key, dial the number to which the call is to be  
transferred. Then press the Send softkey. When the third party phone begins to ring, press  
XFER to complete the transfer.  
NOTE: Call waiting and call transfer must be enabled.  
3.10  
3-way conference call  
1. Press the CONF softkey during an active call.  
2. The first call will be placed on hold and dial tone will be heard.  
3. Dial the number to be added to the conference.  
4. Press Send.  
5. When the call is answered, press CONF to add the caller to the conference.  
6. To release the conference, press SPLIT.  
3.11  
Multiple-way call  
To add a fifth party to four active calls  
1. Press CONF softkey or XFER softkey  
2. Press OK  
3. Enter the number  
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4. Press Send and wait for the other party to answer.  
5. Use the arrow keys to select a call.  
4
Advanced Functions  
4.1  
Call pickup  
This allows a third party to answer a call by dialing a code. For example: A calls B, but  
there is no answer. C can go off hook, dial a code plus B’s number, and pick up the call.  
The following chart shows how to configure this in the dial peer screen.  
*1* is the code. After saving the above configuration, C can dial *1* plus Bs phone number  
to pick up As call. The prefix can be set to anything the user desires that does not interfere  
with other dialing rules.  
4.2  
Join call  
This allows a third party to join an existing call. For example: If B and C are on a call, A can  
join by dialing a code plus the number for B or C. This assumes that B or C also support  
Join Call.  
The following chart shows how to configure this in the dial peer screen.  
*2* is the code. After saving the above configuration, A can dial *2* plus the number for B or  
C to join B and Cs call. The prefix can be set to anything the user desires that does not  
interfere with other dialing rules.  
4.3  
Redial / Unredial  
If B is on a call when A calls, A will get busy tone. If A wants to connect to B as soon as B  
is available, he can use the redial function. To use this feature, A dials a prefix and then B’s  
number.  
When the redial function is activated, A will check B’s calling status every 60 seconds.  
When B is available, A’s phone will ring. When A goes off hook, the phone will call B  
automatically. If A does not want to call B, the redial function can be cancelled by dialing a  
prefix plus B’s number.  
*3* is the redial prefix code. A can dial *3* plus B’s phone number to activate the redial  
function.  
*4* is the unredial prefix code. A can dial *4* to cancel the redial function.  
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The user can select any prefix as long as it does not interfere with dialing rules.  
4.4  
Click to dial  
If User A browses to User B’s phone number or SIP address in the contact page and clicks it,  
User A’s phone will ring. After he goes off hook, the phone will call User B.  
NoteThis feature requires that the PBX support click to dial.  
4.5  
Call back  
This function will redial the last received call.  
4.6  
Auto answer  
If this feature is activated, the phone will answer incoming calls after a programmable delay.  
4.7  
Hotline/Warmline  
This feature will cause the phone to place a call to a programmed number whenever it goes  
off-hook. A different hotline number can be set for each SIP line.  
4.8  
Application  
4.8.1  
SMS  
1. Press Menu ->Applications->Enter->SMS->Enter.  
2. Use the navigation keys to highlight the options. Messages can be read in the  
Inbox/Outbox.  
3. Press Reply to reply to a message. Use the 2aB softkey to change the Input Method.  
After entering the reply, press OK, use the navigation keys to select the line from which  
you want to send, then press Send.  
4. To write a new message, press New. Use the 2aB softkey to change the Input Method.  
After entering the reply, press OK, use the navigation keys to select the line from which  
you want to send, and press Send.  
5. To delete a message, press Del. You have three options to choose: Yes, All, No.  
4.8.2  
Memo  
Memos can be recorded in the phone as reminders.  
Press Menu->Application->Memo->Enter->Add.  
Options for Mode, Date, Time, and Ring Tone can then be configured. The reminder text  
can also be entered. When the configuration is completed, press Save.  
4.8.3  
Voice Mail  
1. Press Menu->Application->Voice Mail->Enter.  
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2. Use the navigation keys to highlight the line for which you want to set voicemail.  
3. Press Edit  
4. Use the navigation keys to enable voicemail.  
5. Input the number. Press 2aB softkey if necessary to change the input method.  
6. Press Save to save the change.  
7. To hear a new voicemail, press the Voicemail softkey. Then press Dial. It may then be  
necessary to enter a password.  
4.8.4  
Ping  
1. Press Menu->Application->Ping->Enter.  
2. Enter the IP Address to be pinged.  
3. Press Start  
4. Display will show “Ping IP Address”  
5. After approximately 5 seconds, the display will show “OK” if the ping is successful or  
“Failed” is the ping is unsuccessful.  
4.9  
Programmable Key Configuration  
The phone has 4 programmable keys which can be set to various functions. The functions are  
discussed in the following sections. The default configuration for each key is None which  
means the key has not been set for any function.  
To configure any function  
1. Press Menu->Settings->Basic Settings->Keyboard->DSS Key Settings.  
2. Choose Line Key Settings or Function Key Settings.  
3. Use the UP ARROW or DOWN ARROW to choose the key.  
4. Use LEFT ARROW or RIGHT ARROW to choose the function.  
4.9.1  
Memory Key  
1. Use the UP ARROW or DOWN ARROW keys to move to Tel.  
2. Enter the number to be stored.  
3. Press SAVE.  
4.9.2  
Line  
Access a SIP or IAX2 line registered to the phone.  
1. Use the UP ARROW or DOWN ARROW keys to move to Line.  
2. Use LEFT ARROW or RIGHT ARROW to select the Line.  
3. Press SAVE.  
4.9.3  
Key Event  
This subtype has many options. They are listed below along with brief explanations.  
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None  
F_MWI Message Waiting  
F_DND Do Not Disturb  
F_HOLD Hold  
F_B_TRANSFER Blind Transfer  
F_PBOOK Phonebook  
F_REDIAL Redial  
F_PICKUP Call Pickup  
F_JOIN Join a call  
F_AUTOREDIAL Auto Redial On  
F_UNAUTOREDIAL Auto Redial Off  
F_CFWD Call Forward  
F_CALLERS Call List  
F_FLASH Flash  
F_MEMO Memo  
F_RELEASE Release Drop call  
F_LOCK Locks the keypad.  
F_SMS Send SMS  
F_HEADSET Activate Headset Mode  
F_LOR Call Back  
F_POWER Turn Power LED On or Off  
F_SDTMF Send DTMF  
F_PREFIX Enter prefix to be dialed. Ex: Access Code for outside line.  
F_HOTDESKING Clears all SIP information and registers new SIP information.  
4.9.4  
DTMF  
Dials a programmed number.  
4.9.5  
URL  
Directly accesses a remote XML phonebook.  
4.9.6  
None  
No function.  
5
Other Functions  
5.1  
Auto Answer  
If this feature is enabled, the phone will answer a ringing line after a specified time.  
1. Press Menu ->Features-> Enter->Auto Answer-> Enter.  
2. Use UP ARROW or DOWN ARROW to select line.  
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3. Use LEFT ARROW or RIGHT ARROW to Enable.  
4. Use UP ARROW or DOWN ARROW to access time setting.  
5. Use keypad to enter time in seconds.  
5.2  
Auto Handdown  
This is the time after a call ends before the phone returns to the idle state.  
1. Press Menu ->Features-> Enter->Auto Handdown-> Enter.  
2. Use LEFT ARROW or RIGHT ARROW to Enable.  
3. Use UP ARROW or DOWN ARROW to access time setting.  
4. Use keypad to enter time in minutes.  
5.3  
Ban Anonymous Call  
If this function is enabled, the phone will block calls with no Caller ID information.  
1. Press Menu ->Features-> Enter->Ban Anonymous Call-> Enter.  
2. Choose the SIP Account from which to Ban Anonymous Call.  
3. Press OK  
4. Use LEFT ARROW or RIGHT ARROW to Enable.  
5.4  
Ban Outgoing  
If this function is enabled, the phone cannot make outgoing calls.  
Press Menu ->Features-> Ban Outgoing-> Enter.  
5.5  
Dial Plan  
1. Press Menu ->Features-> Enter->Dial Plan-> Enter.  
2. The following items in the dial plan can be enabled or disabled: Press # to Send, Timeout  
to Send, Timeout, Fixed Length Number, Press # to Do BXFER, BXFER On Onhook,  
AXFER On Onhook.  
Note: It is recommended that Dial Plan be configured from the web interface.  
5.6  
Dial Peer  
1. Press Menu ->Features-> Enter->Dial Peer-> Enter.  
2. Select Add to enter the Edit interface, and input information.  
Note: It is recommended that Dial Peer be configured from the web interface. Refer to  
Section 8.3.3.4.  
5.7  
Intercom  
Enables/Disables Intercom calls  
Press Menu ->Features-> Enter->Intercom-> Enter.  
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5.8  
Auto Redial  
If Auto Redial is enabled, the phone will continue to retry a busy call. The user sets the retry  
interval and the number of times to redial. The user is also given the option to activate this  
feature on each busy call.  
1. Press Menu ->Features-> Enter->Auto Redial-> Enter.  
2. Use LEFT ARROW or RIGHT ARROW to Enable.  
3. Use UP ARROW or DOWN ARROW to select Interval and Times.  
4. Press Save.  
5.9  
Call completion  
This is similar to Auto Redial except that it detects the state of the called number before  
making a new call attempt.  
1. Press Menu ->Features-> Enter->Call Completion-> Enter.  
2. Use LEFT ARROW or RIGHT ARROW to Enable.  
3. Press Save.  
5.10  
Ring from Headset  
When this function is enabled, ring sound will be passed to a connected headset.  
Press Menu ->Features-> Enter-> Headset Ring -> Enter.  
5.11  
Power Light  
This feature enables the power light at the bottom of the phone.  
Press Menu ->Features-> Enter->Power LED-> Enter.  
5.12  
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.  
1. Press Menu ->Features-> Enter->Hide DTMF-> Enter.  
2. Use LEFT ARROW or RIGHT ARROW to select one of the following 4 choices.  
a) Disabled All the digits will be shown on the LCD.  
b) All – None of the digits will be shown on the LCD. The “*” will be shown.  
c) Delay The last digit entered will be shown for a short time and then replaced by  
“*.”  
d) Last Show The last digit entered will be shown. Previous digits are replaced by  
“*.”  
5.13  
Password Dial  
This feature controls the display of dialed digits. When enabled, a password and length can  
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be set.  
Example: A call is placed to 6625551212. Password is set to 662 and length is set to 3.  
Display will show 662***1212.  
1. Press Menu ->Features-> Enter->Passwd Dial-> Enter.  
2. Use LEFT ARROW or RIGHT ARROW to enable the feature.  
3. Use UP ARROW or DOWN ARROW to move to Prefix.  
4. Use keypad to enter prefix.  
5. Use UP ARROW or DOWN ARROW to move to Length.  
6. Use keypad to enter Length.  
7. Use BACK or EXIT to return to idle screen.  
5.14  
Pre Dial  
If this feature is enabled, digits dialed on-hook will be transmitted when the phone goes  
off-hook  
Press Menu ->Features-> Pre Dial-> Enter.  
6
Basic Setting  
6.1  
Keyboard  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Keyboard->Enter.  
2. There are four sets of keys which can be configured.  
a) DSS Keys Keys on the right side of the phone beside the Speakerphone button or  
Line Keys.  
b) Programmable Keys Arrow keys and OK key  
c) Desktop Long Pressed Action to take when Programmable Key is pressed and  
held.  
d) Soft Key Keys under the display  
3. Use UP ARROW or DOWN ARROW and Enter to select the key.  
4. Use LEFT ARROW or RIGHT ARROW to select the function.  
5. Press OK to save.  
6. Use BACK or EXIT to return to idle screen.  
6.2  
Screen Settings  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Screen Settings->Enter.  
2. The following items can be set.  
a) Contrast Set the contrast of the LCD.  
b) Contrast Calibration Set the level of contrast that the current contrast setting  
provides.  
c) Backlight Enable or disable LCD backlight.  
3. Press OK to save.  
4. Use BACK or EXIT to return to idle screen.  
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6.3  
Ring Settings  
6.3.1  
Ring Volume  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Ring Settings->Enter->Ring  
Volume->Enter.  
2. Use LEFT ARROW or RIGHT ARROW to select the desired ring volume from the 9  
choices. The phone will ring at the selected volume shortly after it is selected.  
3. Press Save.  
4. Use BACK or EXIT to return to idle screen.  
6.3.2  
Ring Type  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Ring Settings->Enter->Ring  
Type->Enter.  
2. Use LEFT ARROW or RIGHT ARROW to select the desired ring type. There are 9  
standard types and 3 user types. The user type can be configured from the web interface.  
The phone will ring at the selected type shortly after it is selected.  
3. Press Save.  
4. Use BACK or EXIT to return to idle screen.  
6.4  
Voice Volume  
1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Voice Volume->Enter.  
2. Use LEFT ARROW or RIGHT ARROW to select the desired voice volume from the 9  
choices.  
3. Press Save.  
4. Use BACK or EXIT to return to idle screen.  
6.5  
Time & Date  
1. Press Menu ->Settings->Enter->Basic Settings-> Enter->Time & Date->Enter.  
2. Use LEFT ARROW or RIGHT ARROW to choose Auto or Manual. If Auto is chosen,  
the phone will get date and time information from a time server. The IP address of this  
server may need to be entered. If Manual is chosen, the date and time must be entered.  
3. Use UP ARROW or DOWN ARROW to move to the following items. Use LEFT  
ARROW or RIGHT ARROW to make selection.  
a) SNTP Server Time Server IP address This is the only item that must be  
configured if auto is chosen.  
b) Time Zone This is shown as an offset from GMT.  
c) Format Date Display format.  
d) Type Character used as delimiter in date display.  
e) 12 Hour Clock If disabled, clock is 24 hour.  
f) Daylight Saving Time  
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4. Press Save.  
5. Use BACK or EXIT to return to idle screen.  
6.6  
Greeting Words  
This feature shows the words displayed in the upper left of the LCD. Default  
is VOIP PHONE.  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Greeting Word->Enter.  
2. Enter the message using the keypad. It may be necessary to change the input mode  
using the soft keys. Use DELETE to remove characters and 0 for space. Maximum  
message length is 12 characters.  
3. Press Save.  
4. Use BACK or EXIT to return to idle screen.  
6.7  
Language  
1. Press Menu ->Settings-> Enter->Basic Settings-> Enter->Language Set ->Enter.  
2. Use LEFT ARROW or RIGHT ARROW to choose English or Chinese.  
3. Press Save.  
4. Use BACK or EXIT to return to idle screen.  
7
Advanced Settings  
7.1  
General  
For all the items in this section, Press Menu->Settings->Enter->Advanced settings->Enter,  
and then enter the password. The default password is 123. It can be changed in the web  
interface.  
7.2  
Account  
This allows configuration of SIP account parameters. After selecting one of the three  
available accounts, the following items may be configured.  
7.2.1  
Basic Settings  
1. Display Name Name send in Caller ID  
2. Outbound Proxy - SIP Outbound Proxy IP Address  
3. Registration Enable or disable registration for this account.  
4. Server Address SIP Server IP Address  
5. Server Port SIP Port Default 5060  
6. SIP User SIP User name  
7. Auth User User name for authentication  
8. Auth Password Password for authentication  
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7.2.2  
Advanced Settings  
1. Domain Realm SIP Domain  
2. Dial Without Registered Enable or disable dialing with no SIP registration  
3. Anonymous Privacy Support. Choose RFC3323, RFC3325 or None  
4. DTMF Mode Choose RFC2833, SIP_Info, In-band, or Auto  
5. Use STUN Enable or disable use of STUN Server. If enabled, the IP address of the  
STUN server must be entered.  
6. Local Port Local SIP Port Default 5060  
7. Ring Type Select ring type for this account. See Section 6.3.2.  
8. MWI Number Number for Message Waiting  
9. Pickup Number Code for call pickup  
10. Park Number Code for call park  
11. Join Call Number Code to join a call  
12. Missed Call Logs Enable or disable  
7.2.3  
Service Code  
Sets the codes to be dialed to an IP PBX to enable or disable the following functions.  
1. Mode Selects whether or not all these codes are active.  
2. DND  
3. Always CFW Always Call Forward  
4. Busy CFW - Call Forward Busy  
5. No Answer CFW - Call Forward No Answer  
6. Anonymous  
7.3  
Network  
Enter Network settings as discussed in Section 2.2.  
7.4  
Security  
1. Menu Password Password to enter configuration menu.  
2. Keyboard Password If this feature is enabled, this password must be entered whenever  
the keypad is used.  
3. Keyboard Status Enable or disable key lock as described above.  
7.5  
Maintenance  
See Section 8.3.6 for a detailed explanation of each option. It is recommended that these  
features be accessed through the web interface.  
1. Auto Provision Select DHCP Option, Plug and Play, or Phone Flash for autoprovision.  
2. TR069 Enable or disable configuration via TR069.  
3. Backup Select Config, Phonebook or none for backup. File name must be entered.  
4. Upgrade Select Image, MMI Set, BMF, Ring, Config, or Phonebook for upgrade. File  
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name must be entered.  
7.6  
Factory Reset  
Choose Yes to return the phone to factory default settings.  
8
Web Configuration  
Introduction of configuration  
Configuration Methods  
8.1  
8.1.1  
There are three methods which can be used to configure this phone:  
Phone keypad As discussed in previous sections  
Web browser - Recommended way  
Telnet with CLI command  
8.1.2  
Password Configuration  
There are two levels of access: root level and general level. A user with root level access  
can browse and set all configuration parameters, while a user with general level can set all  
configuration parameters except server parameters for SIP or IAX2.  
Default user with general level:  
. Username: guest  
. Password: guest  
Default user with root level:  
. Username: admin  
. Password: admin  
The default password for the phone screen menu is 123.  
8.2  
Setting via web browser  
Enter the phone’s IP address into the address bar of the web browser. This assumes that the pc  
and the phone are on the same subnet. Note: Internet Explorer, Firefox, Chrome, or Safari  
are supported browsers.  
If the IP address is not known, it can be displayed on the phone’s LCD by pressing the  
Menu->Status.  
After entering the IP address, the following screen is displayed.  
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After configuring the IP phone, remember to click SAVE under the Maintenance tab. If this  
is not done, the phone will lose the modifications when it is rebooted.  
8.3  
Configuration via WEB  
BASIC  
8.3.1  
8.3.1.1 Status  
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Field Name  
Network  
Explanation  
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).  
Shows the phone numbers and registration status for the 2 SIP LINES  
and 1 IAX2 server.  
Accounts  
8.3.1.2 Wizard  
Select the appropriate network mode. The phone supports three network modes:  
1
2
Static: The parameters of a Static IP connection must be provided by your ISP.  
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.  
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8.3.1.2.4 Quick SIP Settings  
Field Name  
Display Name  
Explanation  
The name shown in caller ID.  
Server Address  
SIP server address either IP address or URI.  
SIP server port (usually 5060).  
Login name or Authentication ID.  
SIP password.  
Server Port  
Authentication User  
Authentication 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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8.3.1.3 Call Log  
Outgoing call logs can be seen on this page.  
Field Name  
Explanation  
Start time of the outgoing call  
Duration of the outgoing call.  
Account, protocol, and line of the outgoing call.  
Start Time  
Duration  
Dialed Calls  
8.3.1.4 Language  
Field name  
Explanation  
Language  
Set the language of phone. English is default.  
The greeting displayed on LCD when phone is idle. It has a  
maximum of 12 characters. Default is VOIP PHONE.  
Greeting Words  
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8.3.2  
Network  
8.3.2.1 WAN Config  
Field Name  
Explanation  
The current IP address of the phone.  
Active IP Address  
Current Subnet Mask  
Current IP Gateway  
MAC Address  
The current Subnet Mask.  
The current Gateway IP address.  
The MAC address of the phone.  
Time the MAC address was obtained.  
WAN Settings  
MAC Timestamp  
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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8.3.2.1.1 Static IP  
If Static IP is chosen, the screen below will appear. Enter values provided by the ISP.  
8.3.2.1.2 DHCP  
If DHCP is chosen, all configuration information will be provided by a DHCP server. Contact  
the ISP to determine if DHCP is used.  
8.3.2.1.3 PPPoE  
If PPPoE is chosen, the screen below will appear. Enter the information provided by the ISP.  
Service Name  
User  
IP Address or name of DSL Server  
DSL User Name or Login ID  
DSL Password  
Password  
After entering the new settings, click the APPLY button. The phone will save the new  
settings and apply them. If a new IP address was entered for the phone, it must be used to  
login to the phone after clicking the APPLY button.  
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8.3.2.2 LAN Config  
Field Name  
Explanation  
IP Address  
LAN static IP.  
Subnet Mask  
DHCP Service  
LAN Subnet Mask.  
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.  
8.3.2.3 Qos & VLAN Config  
The phone supports 802.1Q/P protocol and DiffServ configuration. Use of a Virtual LAN  
(VLAN) allows voice and data traffic to be separated.  
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Chart 1 shows a network switch with no VLAN. Any broadcast frames will be transmitted  
to all other ports. For example, and frames broadcast from Port 1 will be sent to Ports 2, 3,  
and 4.  
Chart 2 shows an example with two VLANs indicated by red and blue. In this example,  
frames broadcast from Port 1 will only go to Port 2 since Ports 3 and 4 are in a different  
VLAN. VLANs can be used to divide a network by restricting the transmission of broadcast  
frames.  
Note: In practice, VLANs are distinguished by the use of VLAN IDs.  
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Field Name  
Enable LLDP  
Explanation  
Enable or Disable Link Layer Discovery Protocol (LLDP)  
The time interval for sending LLDP Packets  
Packet Interval  
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)  
Specify the value of the SIP DSCP in decimal  
Specify the value of the Audio DSCP in decimal  
Enable or Disable WAN Port VLAN  
SIP DSCP  
Audio DSCP  
Enable 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  
Specify the value of the signal 8021.p priority. Range is 0-7  
Follow WAN: LAN Port ID is same as WAN ID  
Disable: Disable Port VALN  
Audio 8021P Priority  
LAN Port VLAN Mode  
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  
0-4095  
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8.3.2.4 Service Port  
Set the port values for Telnet/HTTP/RTP on this page.  
Field Name  
Explanation  
Web Server Type  
HTTP Port  
Specify Web Server Type HTTP or HTTPS  
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.  
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.  
HTTPS Port  
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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8.3.2.5 DHCP SERVICE  
Field Name  
Explanation  
IP-MAC mapping table. If the LAN port of the phone connects to a  
device, this table will show its IP and MAC address.  
Name of the lease table.  
DHCP Client Table  
Leased Table Name  
Start IP Address  
End IP Address  
Beginning IP address of the lease table.  
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 of the lease table.  
Subnet Mask  
IP Gateway  
Leased Time  
DNS Server Address  
Add  
Network Gateway of the lease table.  
Time IP address assignments will persist. Unit is minutes.  
IP address of DNS server.  
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  
Notes:  
Activates DNS Relay in the phone. Default is enabled.  
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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8.3.2.6 TIME&DATE  
Set the time zone and SNTP (Simple Network Time Protocol) server on this page. Daylight  
savings time configuration and manual time and date entry are also done on this page  
Field Name  
Enable SNTP  
Explanation  
Simple Network Time Protocol (SNTP) Settings  
Enable or Disable SNTP  
Enable DHCP Time  
Primary Server  
Secondary Server  
Time Zone  
If this is enabled, phone will synchronize time with DHCP server.  
IP address of Primary SNTP Server  
IP address of Secondary SNTP Server  
Local Time Zone  
Resync Period  
12 -Hour Clock  
Time between resync to SNTP server. Default is 60 seconds.  
If checked, clock is 12 hour mode. If unchecked, 24 hour mode.  
Default is 24 hour mode.  
Date Format  
Specify the date format. Fourteen different formats are available.  
Four date separators are available: /, - , . , space  
Date Separator  
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Daylight Saving Time Settings  
Enable daylight saving time.  
DST offset. Default is 60 minutes.  
Start and end month for DST  
Start and end week for DST  
Start and end day for DST  
Enable  
Offset(minutes)  
Month  
Week  
Day  
Hour  
Start and end hour for DST  
Start and end minute for DST  
Manual Time Settings  
Minute  
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.  
8.3.3  
VOIP  
8.3.3.1 SIP Configuration  
Configure a SIP server on this page.  
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Field Name  
Explanation  
Choose the sip line to configured (SIP 1 SIP 2). Click the dropdown arrow to select the line.  
Status  
Shows registration status. Will show “Registered” if registered  
or “Unapplied” if not registered.  
Server Address  
Server Port  
SIP server IP address or URI.  
SIP server port. Default is 5060.  
Authentication User  
SIP account name (Login ID).  
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.  
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Display Name  
Set the display name. This name is shown on Caller ID.  
Check to submit registration information.  
SIP Domain if different than the SIP Registrar Server.  
SIP proxy server IP address or URIThis is normally the same as  
the SIP Registrar Server)  
Enable Registration  
Domain Realm  
Proxy Server Address  
Proxy Server Port  
Proxy User  
SIP Proxy server port. Normally 5060.  
SIP Proxy server account.  
Proxy Password  
Backup Server Address  
SIP Proxy server password.  
Backup SIP Server Address or URI (This server will be used if the  
primary server is unavailable)  
Backup Server Port  
Server Name  
Backup SIP Server Port  
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.  
DisabledNo call forwarding Default mode  
BusyIf the phone is busy, incoming calls will be forwarded.  
No answerIf there is no answer, incoming calls will be forwarded  
after a specified time.  
AlwaysAll incoming calls will be forwarded.  
Number to which calls are to be forwarded.  
Used in conjunction with Call Forward No Answer. Wait time in  
seconds before call is forwarded.  
Forward Number  
No Ans. Fwd Wait Time  
Transfer Timeout  
Enable Hotline  
Time interval between sending “bye” message and hanging up  
after the phone transfers a call.  
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  
RTP Encryption  
SIP Encryption key.  
Enable/Disable RTP Encryption.  
RTP Encryption Key  
Enable Auto Answer  
RTP encryption key  
Activate Auto Answer mode. If activated, phone will  
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  
If enabled, this will refresh the SIP session timer per RFC4028.  
Refresh interval if Session Timer is enabled.  
Enable Session Timer  
Session Timeout  
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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.  
Time interval between MWI Subscribe Messages.  
Subscribe Period  
Conference Type  
Conference Number  
Choose Conference Type, either local or network  
Number to dial to access network conference server. Not needed  
if Local conference mode is chosen  
Registration Expires  
Enable Service Code  
DND On Code  
SIP re-registration time. Default is 3600 seconds. If the server  
requests a different time, the phone will change to that value.  
Enables or disables the services described below. These codes  
will be sent to the SIP server to activate or deactivate the service.  
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.  
Disable Server Always CFwd as described above.  
Always CFwd Off Code  
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.  
Disable Server Busy CFwd as described above.  
Busy CFwd Off Code  
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.  
Anonymous On Code  
Anonymous On When this function is enabled, the server will  
allow the phone to make anonymous calls. In other words  
“Anonymous” will be transmitted for Caller ID.  
Anonymous Off Code  
Keep Alive Type  
Disable Anonymous Calling function described above.  
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  
server every NAT Keep Alive Period.  
Keep Alive Interval  
User Agent  
Set the NAT Keep Alive Interval. Default is 60 seconds  
Set SIP User Agent value.  
DTMF Type  
DTMF sending mode. There are four modes:  
In-band (Relay)  
RFC2833  
SIP_INFO  
AUTO  
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Different VoIP Service providers may require different modes.  
SIP port. Default is 5060.  
Local port  
Ring type  
Set ring tone. There are 9 standard options and 3 user options.  
Enable/Disable support for NAT traversal via RFC3581 (Rport).  
Enable or disable SIP PRACK function. Default is OFF. It is  
suggested this be used.  
Enable Rport  
Enable PRACK  
Enable Long Contact  
Convert URI  
Allow more parameters in contact field per RFC 3840  
Converts # to %23 when sending URI information.  
Allow outgoing calls without registration.  
Dial Without Registered  
Ban Anonymous Call  
Enable DNS SRV  
Refuse Anonymous Calls  
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  
Server Type  
Enable/Disable BLF List  
Configures phone for unique requirements of selected server.  
Select SIP protocol version RFC3261 or RFC2543. Default is  
RFC3261. Used for servers which only support RFC2543.  
Set transport protocol TCP, UDP or TLS.  
RFC Protocol Edition  
Transport Protocol  
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 serverit will use the source IP address, not the  
address in via field.  
Enable GRUU  
Enable Displayname  
Quote  
Support for Globally Routable User-Agent URI (GRUU)  
Puts quotation marks around the display-name in SIP messages.  
For servers that require this.  
Enable user=phone  
Sets user=phone in SIP messages. For compatibility with servers  
that require this.  
Click to Talk  
Strict Branch  
Set click to Talk (needs support from server).  
SIP Global Settings  
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  
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Registration Failure Retry Registration failure retry time If registration fails, the phone will  
Time  
attempt to register again after registration failure retry time.  
This will affect all lines  
8.3.3.2 IAX2  
Field Name  
Explanation  
Shows registration status. Will show “Registered” if registered  
or “Unapplied” if not registered.  
Status  
Server Address  
Server Port  
IAX2 server address.  
IAX2 server port. Default is 4569.  
Account  
IAX2 account name for registration  
IAX2 registration password.  
Password  
Phone Number  
Local Port  
IAX2 phone number (usually the same as IAX2 account name).  
IAX2 local port. Default is 4569.  
Voice Mail Number  
Voice Mail Text  
Echo Test Number  
Voice mail number.  
Voice mail name.  
If the IAX2 server supports echo test and the echo test number is  
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non- numeric, this number can be used to replace the echo test  
text. This allows dialing a number to perform an echo voice test.  
This function is provided to test whether communication through  
the server.  
Echo Test Text  
Refresh Time  
Echo test text  
Expiration time of IAX2 server registration. Allowed values are  
between 60 and 3600 seconds.  
Enable Registration  
Enable G.729AB  
Enable/Disable IAX2 registration.  
Enable/Disable G.729 codec.  
8.3.3.3 STUN Config  
STUN support is configured in this page.  
STUN Simple Traversal of UDP through NAT A STUN server allows a phone in a private  
network to know its public IP and port as well as the type of NAT being used. The phone can  
then use this information to register itself to a SIP server so that it can make and receive calls  
while in a private network.  
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Field Name  
STUN NAT Transversal  
Server Address  
Explanation  
Shows whether or not STUN NAT Transversal was successful.  
STUN Server IP address  
Server Port  
STUN Server Port Default is 3478.  
STUN blinding period STUN packets are sent at this interval  
to keep the NAT mapping active.  
Binding Period  
SIP Waiting Time  
Waiting time for SIP. This will vary depending on the  
network.  
SIP Line Using STUN  
SIP Line Using STUN  
Use STUN  
Select the Line for use with STUN (SIP 1 - SIP 2)  
Enable/Disable STUN on the selected line.  
8.3.3.4 DIAL PEER  
This feature allows the user to create rules to make dialing easier. There are several different  
options for dial rules. The examples below will show how this can be used.  
Example 1: Substitution Assume that it is desired to place a direct IP call to IP address  
192.168.119. Using this feature, 156 can be substituted for 192.168.1.119.  
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Example 2: Substitution To dial a long distance call to Beijing requires dialing area code  
010 before the local phone number. Using this feature 1 can be substituted for 010. For  
example, to call 62213123 would only require dialing 162213123 instead of 01062213123.  
Example 3: Addition Two examples are shown. In the first case, it is assumed that 0 must  
be dialed before any 11 digit number beginning with 13. In the second case, it is assumed  
that 0 must be dialed before any 11 digit number beginning with 135, 136, 137, 138, or 139.  
Two different special characters are used.  
x Matches any single digit that is dialed.  
[] Specifies a range of numbers to be matched. It may be a range, a list of ranges separated  
by commas, or a list of digits.  
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Field Name  
Phone number  
Explanation  
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.  
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  
Set the Signaling port, the default is 5060.  
Destination  
Port  
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  
Suffix  
Select either SIP or IAX2 protocol.  
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.  
Dial Peer Examples  
Web Interface  
Explanation  
Example  
Set phone number,  
Dial “93333”  
The SIP2 server will  
receive “3333”  
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.  
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  
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The phone will add the alias to Dial “8309“  
the end of the dialed number if The SIP1 server will  
the dialed number matches the receive “07558309”  
template in the Phone Number  
box.  
Set Phone Number, Alias and  
Delete Length. Phone number  
Dial “0106228”  
The SIP1 server will  
is XXXT and Alias is rep: xxx receive “86106228”  
If the dialed phone number  
starts with the digits in the  
Phone Number box, the  
matching digits will be  
replaced by the alias number.  
If the dialed phone number  
starts with the digits in the  
Dial “147”  
The SIP1 server will  
Phone Number box, the phone receive “1470011”  
will send out the dialed phone  
number and add the suffix  
number.  
8.3.4  
Phone  
8.3.4.1 AUDIO  
This page configures audio parameters such as voice codec, handset volume, and ringer  
volume.  
Field Name  
First Codec  
Second Codec  
Explanation  
The first codec choice: G.711A/u, G.722, G.723, G.729, G.726  
The second codec choice: G.711A/u, G.722, G.723, G.729, G.726,  
None  
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Third Codec  
Fourth Codec  
Fifth Codec  
Sixth codec  
Onhook Time  
The third codec choice: G.711A/u, G.722, G.723, G.729, G.726,  
None  
The forth codec choice: G.711A/u, G.722, G.723, G.729, G.726,  
None  
The fifth codec choice G.711A/u, G.722, G.723, G.729, G.726,  
None  
The sixth codec choice G.711A/u, G.722, G.723, G.729, G.726,  
None  
Time the handset must be on hook to disconnect a call. Default is  
200ms.  
Default Ring Type  
Handset Input Volume  
Handset Output  
Volume  
Ring Sound There are 9 standard types and 3 User types  
Handset Microphone volume 9 levels  
Handset receiver volume - 9 levels  
Speakerphone Volume Speaker volume in hands free mode - 9 levels  
Ring Volume  
Ringer Volume - 9 levels  
G729 Payload Length  
Tone Standard  
G729 Payload Length Adjusts from 10 60 mSec  
Select tone plan for the country of operation  
Choices are 160/20ms or 320/20ms  
G722 Timestamps  
G723.1 Bit Rate  
Enable VAD  
Choices are 5.3kb/s or 6.3kb/s  
Enable or disable Voice Activity Detection (VAD). If VAD is  
enabled, G729 Payload length cannot be set greater than 20 mSec.  
The RTP Payload type that indicates DTMF. Default is 101  
DTMF Payload Type  
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8.3.4.2 FEATURE  
This page configures various features such as Hotline, Call Transfer, Call Waiting,  
etc.  
Field Name  
Explanation  
Do Not Disturb  
If enabled, the phone will reject incoming calls. The callers receive  
busy tone. Outgoing calls may be made.  
Enable Call Transfer  
Semi-Attended  
Transfer  
If enabled, Call Transfer is allowed.  
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.  
Wait time before the phone performs the Auto Handdown behavior  
described above.  
Auto Handdown Time  
Enable Auto Redial  
Auto Redial Interval  
If enabled, the phone will automatically redial a call if a busy tone is  
received.  
Wait time between auto redial attempts in seconds.  
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Auto Redial Times  
Auto Headset  
Maximum number of auto redial attempts.  
Automatically answers call on headset.  
Enable Intercom  
Enable Intercom Tone  
P2P IP Prefix  
If enabled, allows intercom calls.  
If enabled, plays intercom ring tone to alert to an intercom call.  
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  
Emergency Call  
Number  
Disables Power Light if selected.  
The phone will dial the emergency call number even if the keyboard  
is locked.  
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.  
Prefix for password dialing as described above.  
Length for password dialing as described above.  
If enabled, no outgoing calls can be made.  
Password Dial Prefix  
Password Dial Length  
Ban Outgoing  
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  
This is similar to Auto Redial except that the phone detects the state  
of the called number before making a new call attempt.  
If this feature is enabled, digits dialed on-hook will be transmitted  
when the phone goes off-hook.  
Completion  
Enable Pre-Dial  
Enable Silent Mode  
Hide DTMF  
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.  
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 “*”  
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 “*.”  
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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  
Reject Return Code  
Specify SIP Code returned for Busy. Default is 486 Busy Here.  
Specify SIP Code returned for Rejected call. Default is 603 –  
Decline.  
Active URI Limit IP  
Push XML Server  
IP address of the server for the Action URL messages described  
below.  
IP address for XML server which can send display content to the  
phone.  
Enable Call Waiting  
Tone  
Enables audible notification of call waiting.  
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  
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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8.3.4.3 DIAL PLAN  
This phone supports 7 dialing modes:  
1. End with “#”– Dial the desired number, and press # to send it to the server.  
2. Fixed Length The number will be sent to the server after the specified number of digits  
are dialed.  
3. Time Out Number will be sent to the server after the specified time.  
4. User Defined Customized rules created by the user.  
There is a special feature in the dial plan for the case where the user must dial an access code  
to get an external line. A digit followed by a “,” will cause secondary dial tone to be  
generated. For example, assume a rule “9,xxxxxxx” is added. When the user dials 9, the  
phone will generate secondary dial tone. Then, when 8 digits have been dialed, they will all  
be sent to the server.  
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5. Press # to Do Blind Transfer - Press # after entering the target number for the transfer.  
The phone will transfer the current call to the third party.  
6. Blind Transfer on Onhook - Hang up after entering the target number for the transfer. The  
phone will transfer the current call to the third party.  
7. Attended Transfer on Onhook - Hang up after the third party answers. The phone will  
transfer the current call to the third party.  
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.  
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.  
Cause extensions 1000-8999 to be dialed immediately  
Cause 8 digit numbers beginning with 9 to be dialed immediately  
Cause 911 to be dialed immediately  
Cause 99 to be dialed after 4 seconds.  
Cause any number beginning with 9911 to be dialed 4 seconds after dialing ceases.  
Note: End with “#”, Fixed Length, Time out and Digital Map Table can be used  
simultaneously.  
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8.3.4.4 CONTACT  
Enter the name, phone number and ring type for each contact here.  
Field Name  
Explanation  
Phonebook Tables  
Dropdown box to select group  
Contact name  
Group  
Name  
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Office Number, Mobile  
Number, Other Number  
Ring Type  
Contact phone numbers  
Ring type for this contact  
Contact group for this contact  
Add Contact  
Group  
Name  
Contact name  
Office Number, Mobile  
Number, Other Number  
Line  
Contact phone numbers  
Select line for associated contact number  
Ring type for this contact  
Ring Type  
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 CSV  
Export VCF  
Export contacts to xml file.  
Export contacts to csv file.  
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  
Value  
Line  
Select the blacklist type - number or prefix  
Input number or prefix  
Select the sip line  
Note: The maximum capability of the phonebook is 500 contacts.  
Note: “x” and “.” are special characters in the black list. “x” matches any single digit and “.”  
matches any number of digits. For example, 4xxxmatches 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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8.3.4.5 REMOTE CONTACT  
Allows access to remote contact lists either via XML or LDAP.  
TFTP example: Set the Phonebook Name as cortelco - Server URL is  
LDAP example: Server URL is ldap://192.168.1.3/dc= winline,dc=com.  
Remote Phonebook Setting  
Phonebook Name  
Server URL  
Phonebook name displayed on the phone.  
Server url of the remote phonebook.  
SIP line for the remote phonebook.  
SIP Line  
Authentication  
User/password  
Authentication mode for remote phonebook.  
Authentication username and password.  
8.3.4.6 WEB DIAL  
This feature allows a call to be initiated by a computer. To place a call, enter the number in  
the Dial Number box, select the line in the Line Selection box and press the Dial button. To  
end the call, press the Hangup button.  
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8.3.5  
Function Key  
The phone has 4 programmable DSS/Function keys with associated LEDs. The 4 directional  
arrow keys and the OK button are also programmable. This screen also sets the LCD contrast  
and enables the backlight.  
8.3.5.1 Function Keys  
Screen Configuration  
Field Name  
Contrast  
Explanation  
Set screen contrast  
Enable Backlight  
Enable/disable LCD backlight.  
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.  
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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 on which the function is to be performed.  
Used with Key Event and Memory Key. Further specifies the type  
of function to perform.  
Line  
Subtype  
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  
8.3.5.2 Softkeys  
Configure the functions performed by the softkeys under the LCD in various phone operating  
modes.  
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8.3.6  
Maintenance  
8.3.6.1 Auto Provision  
The phone supports PnP, DHCP, and Phone Flash to obtain configuration parameters. They  
will be queried in the following order when the phone boots.  
DHCP PnP server Phone Flash  
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  
User  
Serial number of the phone  
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 for common configuration file  
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Encryption Key  
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 option number. Must be from 128 to 254.  
Custom DHCP Option  
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 Port  
PnP Server Address  
PnP Server Port  
PnP Transport  
PnP Interval  
PnP Transfer protocol UDP or TCP  
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.  
Specify the Protocol type FTP, TFTP or HTTP.  
Specify configuration file name. The phone will use its MAC ID  
as the config file name if this is blank.  
Protocol Type  
Config File Name  
Update Interval  
Update Mode  
Specify the update interval time. Default is 1 hour.  
1. Disable no update  
2. Update after reboot update only after reboot.  
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3. Update at time interval update at periodic update interval  
TR069 Settings  
Enable TR069  
Enable/Disable TR069 configuration  
Select Common or CTC ACS Server Type.  
ACS Server URL.  
ACS Server Type  
ACS Server URL  
ACS User  
User name for ACS.  
ACS Password  
ACS Password.  
TR069 Auto Login  
"Inform" Sending Period  
Enable/Disable TR069 Auto Login.  
Time between transmissions of “Inform” Unit is seconds.  
8.3.6.2 Syslog  
Syslog is a protocol used to record log messages using a client/server mechanism. The  
Syslog server receives the messages from clients, and classifies them based on priority and  
type. Then these messages will be written into a log by rules which the administrator has  
configured. There are 8 levels of debug information.  
Level Name  
Description  
0
1
2
3
4
Emergency  
System is unusable. This is the highest debug info level.  
Action must be taken immediately.  
Alert  
Critical  
Error  
Critical conditions. System is probably working incorrectly.  
Error conditions. System may not work correctly.  
Warning conditions. System may work correctly but needs  
attention.  
Warning  
5
6
7
Notice  
Normal but significant condition.  
Informational Normal daily messages.  
Debug Debug messages normally used by system designer. This level  
can only be displayed via telnet.  
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Syslog Configuration  
Field Name  
Syslog Settings  
Server IP  
Explanation  
Syslog server IP address.  
Server Port  
Syslog server port.  
MGR Log Level  
SIP Log Level  
IAX2 Log Level  
Enable Syslog  
Web Capture  
Start  
Set the level of MGR log.  
Set the level of SIP log.  
Set the level of IAX2 log.  
Enable or disable syslog.  
Capture a packet stream from the phone. This is normally used to  
troubleshoot problems.  
Stop  
Stop capturing the packet stream  
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8.3.6.3 Config Setting  
Config Setting  
Field Name  
Explanation  
Save Configuration  
Save the current phone configuration. Clicking this saves all  
configuration changes and makes them effective immediately.  
Save the phone configuration to a txt or xml file. Please note to  
Right click on the choice and then choose “Save Link As.”  
Logged in as Admin, this will restore factory default and remove all  
configuration information.  
Backup Configuration  
Clear Configuration  
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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8.3.6.4 Update  
This page allows uploading configuration files to the phone.  
Update  
Field Name  
Explanation  
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.  
Web Update  
TFTP/FTP Update  
FTP/TFTP server address for download/upload. The address can be  
IP address or Domain name with subdirectory.  
FTP server Username for download/upload.  
FTP server password for download/upload.  
Name of update file or config file. The default name is the MAC of  
the phone.  
Server Address  
User  
Password  
File name  
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.  
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Type  
Action to be executed by the phone.  
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  
Select File  
Logo File  
URL of the logo file.  
Delete Logo File  
Logo file name to be deleted.  
Logo File  
Logo file in use.  
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8.3.6.5 Access  
User accounts can be added or deleted from this page. The authority of accounts can also be  
changed.  
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.  
Set the password  
Password  
Confirm  
Confirm the password  
User Management  
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.  
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8.3.6.6 Reboot  
Some configuration modifications require a reboot to become effective. Clicking the Reboot  
button will cause the phone to reboot immediately.  
Note: Be sure to save the configuration before rebooting.  
8.3.7  
Security  
8.3.7.1 WEB FILTER  
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  
Start IP Address  
End IP Address  
Add  
Explanation  
Beginning IP Address for MMI Filter  
Ending IP Address for MMI Filter  
Add this filter range to the Web Filter Table  
Select to enable MMI Filter.  
Enable Web 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.  
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8.3.7.2 Firewall  
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  
Enable Input Rules  
Enable Output Rules  
Input/Output  
Explanation  
Enable rules limiting access from the Internet.  
Enable rules limiting access to the Internet.  
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  
Src Mask  
Set destination address. It can be a single IP address or use * as a  
wild card. For example: 192.168.1.14 or *.*.*.14.  
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.  
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.  
Dest Mask  
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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.  
8.3.7.3 Network Address Translation (NAT)  
NAT is the process of modifying IP address and port information in transition from a private  
to a public network. NAT allows the use of one public address to support many private  
addresses.  
DMZ Configuration  
Servers in a network most vulnerable to attack are those which provide services to users  
outside the local network. Many times these computers are placed into their own  
sub-network to provide more protection to the rest of the local network. This sub-network is  
called a DMZ (taken from “demilitarized zone”). Computers in the DMZ have limited  
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connectivity to specific hosts in the internal network, although communication with other  
hosts in the DMZ and to the external network is allowed. This allows hosts in the DMZ to  
provide services to both the internal and external network, while a firewall controls the traffic  
between the DMZ servers and the internal network clients.  
The following chart describes the network access control of DMZ.  
Application Layer Gateway (ALG) Settings  
Field Name  
Explanation  
IPSec ALG  
Enable/Disable IPSec encryption. Default is enabled.  
Allow the ALG to securely pass FTP traffic. Default is enabled.  
Allow the ALG to securely pass PPTP traffic. Default is enabled.  
FTP ALG  
PPTP ALG  
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Network Address Translation (NAT) Table  
Shows the NAT TCP and UDP mapping tables  
NAT Table Option  
Transfer Type  
Inside IP  
Select the TCP or UDP protocol.  
Set the local IP address of device.  
Inside Port  
Outside Port  
Set the LAN (inside) port for NAT mapping  
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.  
8.3.7.4 VPN  
The phone supports remote connection via VPN. It supports both Layer 2 Tunneling  
Protocol (L2TP) and OpenVPN protocol. This allows users at remote locations on the  
public network to make secure connections to local networks.  
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  
L2TP  
Enable/Disable VPN.  
Select Layer 2 Tunneling Protocol  
Select OpenVPN Protocol  
OpenVPN  
Only one protocol may be activated. After the selection is made, the configuration should be  
saved and the phone rebooted.  
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VPN Server Address  
VPN User  
Set VPN L2TP Server IP address.  
Set User Name access to VPN L2TP Server.  
Set Password access to VPN L2TP Server.  
VPN Password  
8.3.7.5 Security  
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  
HTTPS File  
Show SIP TLS authentication certificate.  
Show HTTPS authentication certificate.  
Show OpenVPN File authentication certificate file.  
OpenVPN Files  
8.3.8  
Logout  
Click Logout to exit the phone web page.  
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9
Appendix  
9.1  
9.1.1  
Specification  
Hardware  
Item  
Specification  
Power Adapter  
Input: 100-240V  
Output: 5V 1A  
WAN  
10/100Base- T RJ-45 1 PORT  
10/100Base- T RJ-45 1 PORT  
RJ9 1 PORT  
Port  
LAN  
Headset  
Power Consumption  
Idle: 2.5W  
Active: 2.8W  
LCD Size  
128x48  
74x28mm  
Operation Temperature 040  
Relative Humidity  
CPU  
1065%  
Broadcom  
16MB  
SDRAM  
Flash  
4MB  
Dimension(L x W x H) 295×295×175mm  
Weight  
1.5kg  
9.1.2  
Voice Features  
Supports 2 SIP servers  
Supports RFC3261  
Codecs  
. G.711A/u  
. G.723.1 high/low  
. G.729a/b  
. G.722  
. G.726  
. Codec Setting per SIP line  
Echo cancellation: G.168 Compliance in LEC, additional acoustic echo  
cancellation(AEC) can reach 96ms max filter length in hands-free mode  
Supports Voice Gain Setting, VAD, CNG  
Full duplex hands-free  
Multi line HD Voice  
SIP support  
. SIP domain  
Version 1.0  
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. SIP authentication  
none  
basic  
MD5  
DNS  
Peer to Peer/ IP call  
Automatic line selection  
9 Standard ring tones and 3 user-defined ring tones  
DTMF  
. SIP info  
. DTMF Relay (In-Band)  
. RFC2833  
. AUTO  
SIP applications  
. Call Forward  
. Call TransferBlind/Attended)  
. Hold  
. Call Waiting  
. 3 Way Conference  
. SMS  
. Remote Pickup  
. Join Call  
. Redial  
. Unredial  
. Multi-line  
. Intercom  
. BLF  
. Presence  
. Push to talk  
. Auto Redial  
. Call Back  
Call control features  
. Flexible dial plan  
. Hotline  
. Anonymous Call Reject  
. Black List (Reject Authenticated Call)  
. Approved Caller List  
. Limit Call  
. Do Not Disturb  
. Caller ID  
. CLIR (reject anonymous call)  
. CLIP(make anonymous call)  
. Dial without Registration  
Phonebook 500 records  
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. Incoming Calls  
. Outgoing Calls  
. Missed Calls  
. Max of 300 Records Each  
. Supports vCard/XML/CSV  
Support IAX2  
4 DSS keys  
Programmable Soft Keys  
Programmable Function Keys  
Code synchronization  
. IP PBX  
. IMS  
Supports Click to Dial via Web Phone Book  
Keypad Lock with Emergency Call  
Customized LCD logo as screensaver  
Ring Tone via Headset or Speaker  
Customized Signal Tone Parameters  
Time Display  
. 12/24 Hour  
. Support Daylight Saving Time  
Supports Path, Group  
Supports SIP Privacy  
Supports MWI  
Supports Speed Dial  
Supports XML  
9.1.3  
Network Features  
WAN/LAN  
. Bridge  
. Bridge with port mirror  
. Router  
Supports PPPoE for xDSL  
Supports Basic NAT and NAPT  
Supports VLAN  
. 802.1Q  
. 802.1P  
Supports STUN  
Supports DMZ  
Supports VPN  
. L2TP  
. OpenVPN  
Wan Port Supports Main DNS and Secondary DNS  
Supports DNS via DHCP or Static DNS  
Supports DHCP client on WAN  
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Supports DHCP server on LAN  
QoS with DiffServ  
Network Tools in Telnet Server  
. Ping  
. Trace Route  
. Telnet Client  
9.1.4  
Maintenance and management  
Firmware Upgrade  
. POST  
. HTTP  
. FTP  
. TFTP  
. HTTPS  
Configuration  
. Web  
. Telnet  
. Phone Keypad  
Two Account Levels  
Multi-Language Support  
. English  
. Chinese  
. Spanish  
. French  
. Portuguese  
. German  
. Russian  
Supports Syslog  
Supports Auto Provisioning  
. Firmware Upgrade  
. Auto-Provisioning  
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9.2  
Digit-character map table  
Keypad  
Character  
1 @  
Keypad  
Character  
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  
4 G H I g h i  
5 J K L j k l  
9 W X Y Z w x y z  
*/.  
0
6 M N O m n o  
#/SEND  
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