SIP-Phone is a full-featured IP-based telephone set via Ethernet base communication. Over
the office LAN connection, it provides IP-PBX solution such as station-to-station call, IP call
and local PSTN/PBX Extension call via PSTN Gateway.
SIP-Phone provides two 10/100BaseT switch/hub RJ-45 ports allow connecting to office
LAN and PC simultaneously. It is compatible with ADSL or Cable Modem provided by ITSP,
ISP or Carrier Company to provide VoIP services to residential and SOHO application.
SIP-Phone is also an integrated Analog Phone provides IP call or PSTN call selection.
When external power is down, it can be a Plain Old Telephone set (POTs).
It provides internal high-quality speakerphone, programmable keys and feature buttons.
SIP-Phone also embedded with a dot matrix of two lines 24 characters LCD, which can
display date and time, calling party name, calling party number, and digits dialed and etc.
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1. Hardware Overview
1.Front View and Keypad function
♦ LCD: 2 lines, 24 character Dot Matrix display.
♦ C: Jump out current LCD menu.
♦ : Move to previous selection or clear previous data.
♦ : Move to right or next selection.
♦ OK: Press OK to confirm the modification.
♦ Direct Line (DL) Button 1 – 10: User press DL button after off-hook to do speed dial
according to phone book data from 1-10 (please refer to LCD configuration-3.Phone
Book; Advanced Configurations via Telnet- 10.[pbook] command, or Web
Configuration-Phone Book chapter.
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♦ Number 1 –10, * and #: The function is the same as the general phone set.
♦ MUTE: Mute the voice of MIC and let others can’t hear from user in communication.
♦ PSTN: Press PSTN to switch SIP-Phone as PSTN or IP Phone Mode. In PSTN mode,
“PSTN” characters will be displayed on LCD left bottom side, then users can dial out as
if standard telephone set in PSTN; in IP Phone mode, “PROXY ” characters will be
displayed on LCD left bottom side.
Note:
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Phone has both IP and
If in communication with PSTN side, user must hang up PSTN side before pick up IP
If A press TRANSFER and number, press TRANSFER again can cancel
ringing, press TRANSFER
1. When SIP-Phone is in PSTN mode, only PSTN and SPEAKER function key can work.
2. On LCD will display ”…Incoming Call... ” to inform user when SIPPSTN side incoming calls.
3. If in communication with IP side, user can press HOLD to hold IP side, then press PSTN
to pick up PSTN side, after that can press HOLD again to retrieve IP side.
4.
side.
♦ HOLD: To hold a call, after press HOLD button, both sides will hear hold tone.
♦ SPEED:
1. Press SPEED and number (Phone book index) after off-hook can do speed dial
according to phone book data (please refer to 3.LCD configuration-6. Phone Book
or Advanced Configurations via 4.Telnet- 10. [pbook] command).
2. Switch input mode between character mode or digit mode, e.g., when user wants to
input phone number can press SPEED to switch input mode as digit mode; when
user wants to input name can also press SPEED to switch input mode as character
mode.
♦ FORWARD: Forward an incoming call to another IP device. (Please refer to LCD
configuration-Forward Type)
♦ MESSAGE and its indicated LED light: When having missed incoming calls, the
MESSAGE LED will be flashing. User can check the information of missed calls by
pressing the MESSAGE button.
♦ TRANSFER:
1. Transfer a call to the third site. When A and B are in communication, A wants to
transfer this call to C, A can press TRANSFER button, now B will hear hold tone,
and A will hear dial tone, then A can press phone number of C, after C picks up, A
can talk with C, after A hangs up, B and C can be connected.
Note:
1. A cannot press phone number of C before hearing dial tone.
2.
transferring and retrieve call with B.
3. If A press TRANSFER and number of C, C is
again C will return to standby mode and A can retrieve call with B.
4. Before C picks up, A cannot hangs up the phone.
2. Change characters to be capital or lowercase: when pressing TRANSFER before
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press letters can switch input type of letters.
♦ REDIAL: Redial the last outgoing call.
♦ + And -: Adjust the voice volume heard of communication.
♦ SPEAKER: Hand free mode. User can talk without picking up handset.
Note:
1. All function keys mentioned above (except dialing keypad) are effective only in IP
Phone mode.
2. When SIP Phone fail to register to Proxy server under Proxy mode, when user
wants to dial out, SIP Phone will play busy tone, and on LCD will display
“Register Failed.”
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2.Back View
♦ DC 9V: DC 9V power input outlet
♦ LAN: RJ-45 connector, connected directly to the Hub through the straight CAT-5
cable.
♦ PC: RJ-45 connector, connected directly to the PC through the straight CAT-5 cable
♦ Line: RJ-11 connector, connected directly to the PSTN analog line.
Note:
There are two LED indicated lights: LINK/ACT and 10/100 for LAN port and PC port.
When network status is regular, LED of LINK/ACT will light on; when SIP-Phone is
transmitting or receiving data, LED will be flashing; when transmit rate is in 10 mbps or
100mbps, LED of 10/100 will light off or light on.
3.Specification of connector
1、 Ethernet Port:
Ethernet port is for connecting SIP-Phone to network, transmit rate supports 10/100
Base-T.
Ethernet connector(LAN)
2、 RJ-11connector:
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RJ-11 connector is for connecting SIP-Phone with PSTN.
RJ11connector
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2. Software Features and Specification
Application:
ISP/ITSP (Internet Telephony Service Provider)
IP-PBX with office telephony services
Multi-nation enterprise communication
SOHO Telephony
Calling Features
Call Hold
Call Transfer
Call Forward
10 sets last number Redial
Ten configurable speed dials
Network Supported
Fixed IP
Dynamic Host Configuration Protocol (DHCP)
PPPoE connection (When PPPoE disconnect, SIP-Phone can automatically
re-connect)
Behind NAT IP Sharing Device
Support QOS by setting DSCP (Differentiated Service Code Point) parameters of VoIP
SIP (RFC3261) compliance
LCD configuration password protection
Provide Proxy Mode or Peer-to-Peer Mode (Non Proxy Server needed) selection
Ring tone, Speaker and Handset volume adjustable
Dial path selection (PSTN or IP mode)
Support DNS server inquiry
Management Features:
Software Upgrade: TFTP/FTP download
Three easy ways for system configuration
- LCD Front Panel
- Web Browser
- TELNET
Certification
CE, 3C
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3. Physical specification & Environment
Dimension:
215mm(W) x 71mm(H) x 198mm(D)
Weight:
834 grams (unit)
Power Supply:
DC 9V output adaptor, AC 120Vac input
Operation Environment:
Humidity: 10 to 90 % (Non-condensing)
Operational Temperature: 0 to +40 °C
Storage Environment:
Humidity: 10 to 90 % (Non-condensing)
Storage Temperature: -10 to +50 °C
Chapter 2 Configuring the SIP-Phone through LCD Phone
menus
Note:
1. After any configuration is made for the SIP-Phone, user has to do Reboot
selection 7 “Reboot”.
2. We suggest user to set IP address via LCD menu 5→→→→
chapter 3 to do other detail configurations via web browser.
1. Initialize SIP-Phone
1. When power on the SIP-Phone, the LCD screen shows as below. Now SIP-Phone is
running Boot sector program.
IP-Phone
Board Start Booting
2. When SIP-Phone finishes boot program initialization. User can see flashing greeting
as below:
System Initializing…………..
3. Then SIP-Phone get into standby mode:
Proxy 10:10:10 AM
SIP-Phone
The main LCD screen would be shown as similar as above. “Proxy” means the
SIP-Phone is in Proxy Mode, and when SIP-Phone is connected to SNTP server, on
LCD will show current time captured from SNTP server.
4. When SIP-Phone is under peer-to-peer mode, on LCD will show “P2P” instead of
“Proxy”.
5. After pressing the PSTN button, the “
SIP SIP-Phone Administration Guide
SIP-Phone
P2P 10:10:10 AM
Proxy” or “P2P” will be replaced by “PSTN”.
14
Please notice that user must plug PSTN line in RJ-11 port when SIP-Phone is in
PSTN mode. SIP Phone will always stay in IP mode, after a PSTN call is finished,
SIP Phone will automatically return to IP mode..
PSTN 10:10:10 AM
IP-Phone
6. Press or to enter configuration mode then press OK button to enter sub menus;
press C can jump out current menu to previous level.
1. Call List
2. Forward Type
3. Phone Book
4. Ringer Settings
5. Network
6. Advanced Settings (protected by password)
7. Reboot
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2. LCD Menu Configuration
User can set the following configurations by LCD keypad.
Note:
1. Press TRANSFER before input data can switch characters to be capital or
lowercase.
2. Press SPEED before input data can switch input mode as character mode or
digit mode; for example, user wants to enter IP address, after pressing SPEED
can enter digits directly.
3. When user is inputting data, press C will jump out to menu list, press will
clear previous input data.
1. Call List
If there is an unanswered IP call, it will be kept in message box. MESSAGE LED will
be flashing until user press MESSAGE to check miss call and re-press MESSAGE
to return to main screen.
(1) Missed Calls::::to see all missed calls in message box.
(2) Received Calls::::to see all received calls in message box.
(3) Dialed Numbers: to see all dialed calls in message box.
2. Forward Type
There are 3 selections in Forward type, user must select under which condition to
forward calls.
(1) Busy
When SIP-Phone is in busy status, the incoming call will be
forwarded to the assigned phone number.
A. Activate
Enter a forwarded phone number to activate busy forward function.
B. Deactivate
Deactivate Busy Forward function.
(2) No Response
When SIP-Phone hasn’t been picked up for around 10 seconds, the
incoming call will be forwarded to the assigned phone number.
A. Activate
Enter a forwarded phone number to activate no response forward
function.
B. Deactivate
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3. Phone Book
1. List
Deactivate No Response Forward function.
(3) Unconditional
It is included the above two types. Whether the SIP-Phone is in
which status, calls will be automatically forwarded to the assigned
phone number.
A. Activate
Enter a forwarded phone number to activate Unconditional Forward
function.
B. Deactivate
Deactivate Unconditional Forward function.
List all records of name, telephone number, and IP address in the phone
address book.
2. Edit/Delete
Edit or delete a record of name, telephone number, and IP address of the
phone address book.
3. New
Add a new record of name, telephone number, and IP address of the phone
address book.
4. Ringer Settings
1. Volume
User can adjust ring volume by press or on the keypad to decrease or
increase ringer volume.
2. Style
There are three tone styles for SIP-Phone. Move the “>” symbol by press
or on the keypad to select the tone style preferred, then press OK to
confirm it.
5. Network
1. Information
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Phone is under DHCP mode, then change to Static mode,
the following items: IP address, Subnet Mask, Default Gateway, will
User can press or to check current network status:
(1) Mode: Display current network connection mode of SIP-Phone to be
Static (Fixed IP), DHCP, or PPPoE.
Note:
When SIP-
display empty, after reboot, user can see information again.
(2) IP, Mask, Gateway: display current IP information.
2. Network Mode
Set network mode of SIP-Phone to be Static (Fixed IP), DHCP, or PPPoE.
3. IP address
Set IP address of SIP-Phone.
4. Subnet Mask:
Set subnet mask address of SIP-Phone.
5. Default Gateway
Set default gateway address of SIP-Phone.
6. Domain Name Server
Set IP address of Domain Name Server. Once SIP-Phone can connect to
DNS server, user can set URL address for Proxy server or Phone book
instead of IP address.
7. PPPoE Configuration
(1) User Name
Set PPPoE connection authentication user name.
(2) Password
Set PPPoE connection authentication password.
(3) Auto Re-connect
Choose ON or OFF to enable or disable this function. If user enables this
function, after PPPoE disconnected, SIP-Phone will automatically reboot
to re-connect, and after reboot, if SIP-Phone still can’t connect with server,
SIP-Phone will keep trying to connect. On the other hand, if user disables
this function, SIP-Phone won’t reboot and keep trying to connect.
8. SNTP Configuration
(1) SNTP Mode:
User can set SNTP function to be on or off, which means SIP-Phone will
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menu directly, once
, user must input
the distributor, we will
generate a specific password according to your MAC address of
Phone via Telnet or Web browser with
default IP address: 10.1.1.3. (This only works when default IP address
capture current time from SNTP server or not.
(2) SNTP Server:
User can specify a SNTP server for SIP-Phone to capture current time.
(3) Time Zone:
User can set time zone via pressing or according to the location
SIP-Phone is. For example, in Taiwan the time zone should be set as
GMT+8:00.
9. Behind IP-Sharing
(1) If SIP-Phone is behind IP sharing or NAT device, on IP sharing must
enable “DMZ” function or set “Virtual Server” to open ports (UDP port:
5060 and 16384).
(2) User must enter public IP address of IP sharing.
6. Advanced Settings (protected by password)
Please Enter Password:
User must key in password to enter this menu, selections under this
command are all important ones, which can only be configured by advanced
users.
Note:
1. Default Password is empty, user can enter this sub-
password has been set in 3.LCD Menu password
password before enter this sub-menu.
2. If user forget password, please contact with
SIP-Phone.
3. User can also try to configure SIP-
hasn’t been changed.)
1. SIP Settings
(1) Connect Mode
Select SIP connection mode to be peer-to-peer mode or Proxy mode.
(2) Proxy
A. Proxy
Set Proxy IP address or Domain Name.
B. Outbound proxy
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