SVP309 Home SIP Phone User Manual

SVP309 Home SIP Phone User Manual
SVP309 Home SIP Phone User Manual
Version 1.2
STEPHEN TECHNOLOGIES CO.,LIMITED
ALL RIGHTS RESERVED
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
SVP309
9 Home SIP Phone User Manual
Safety Notices
Please read the following safety notices before installing or using this phone. They are crucial for the safe and reliable
operation of the device.
Please use the external power supply that is included in the package. Other power supplies may cause damage to the
phone, affect the behavior or induce
uce noise.
Before using the external power supply in the package, please check with home power voltage. Inaccurate power
voltage may cause fire and damage.
Please do not damage the power cord. If power cord or plug is impaired, do not use it, it may cause
cau fire or electric
shock.
socket combination must be accessible at all times because it serves as the main disconnecting device.
The plug-socket
Do not drop, knock or shake it. Rough handling can break internal circuit boards.
Do not install the device in places where there is direct sunlight. Also do not put the device on carpets or cushions. It
may cause fire or breakdown.
Avoid exposure the phone to high temperature, below 0
Do not attempt to open it. Non-expert
expert handling of the device could damage it. Consult your authorized dealer for
help, or else it may cause fire, electric shock and breakdown.
Do not use harsh chemicals, cleaning solvents, or strong detergents to clean it. Wipe it with a soft
so cloth that has been
slightly dampened in a mild soap and water solution.
When lightning, do not touch power plug or phone line, it may cause an electric shock.
Do not install this phone in an ill-ventilated
ventilated place.
You are in a situation that could cause
use bodily injury. Before you work on any equipment, be aware of the hazards
involved with electrical circuitry and be familiar with standard practices for preventing accidents.
℃ or high humidity. Avoid wetting the unit with any liquid.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
2
SVP309 Home SIP Phone User Manual
Table of Content
1. Introducing SVP309 VoIP Phone ............................................................................................5
1.1. Thank you for your purchasing SVP309............................................................................. 5
1.2. Delivery Content ................................................................................................................ 5
1.3. Keypad ............................................................................................................................... 5
1.4. Port for connecting............................................................................................................ 7
2.Initial connecting and Setting ....................................................................................................7
2.1. connect the phone ............................................................................................................ 7
2.2. Initial Setting...................................................................................................................... 9
2.2.1. PPPoE mode. ............................................................................................................... 9
2.2.2. Static IP mode: .......................................................................................................... 10
2.2.3. DHCP mode ............................................................................................................... 11
3. Basic Functions........................................................................................................................11
3.1. Basic operation ................................................................................................................ 11
3.1.1. Accepting a call ......................................................................................................... 11
3.1.2. Making a call ............................................................................................................. 11
3.1.3. Ending a call .............................................................................................................. 12
3.1.4. Transferring a call ..................................................................................................... 13
3.1.5. Calling Hold and 3 ways call ...................................................................................... 13
3.1.6. Callers........................................................................................................................ 13
3.2. The high-level operation ................................................................................................. 14
3.2.1. Special Keys ............................................................................................................... 14
3.2.2. Call pickup ................................................................................................................. 15
3.2.3. join call ...................................................................................................................... 15
3.2.4. redial/unredial .......................................................................................................... 15
3.2.5. click to dial ................................................................................................................ 16
4. Setting .....................................................................................................................................16
4.1. Setting methods .............................................................................................................. 16
4.2. Setting via Web Browse .................................................................................................. 16
4.3. Configuration via WEB..................................................................................................... 17
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
3
SVP309 Home SIP Phone User Manual
4.3.1. BASIC ......................................................................................................................... 17
4.3.1.1. Status .................................................................................................................. 17
4.3.1.2. Wizard ............................................................................................................... 17
4.3.1.3. Call Log ............................................................................................................... 20
4.3.1.4. MMI SET ............................................................................................................ 20
4.3.2. Network .................................................................................................................... 20
4.3.2.1. WAN Config ........................................................................................................ 20
4.3.2.2. LAN Config .......................................................................................................... 22
4.3.2.3. Qos Config .......................................................................................................... 23
4.3.2.4. Service Port ........................................................................................................ 25
4.3.2.5. DHCP SERVER ..................................................................................................... 26
4.3.2.6. SNTP ................................................................................................................... 28
4.3.3. VOIP .......................................................................................................................... 29
4.3.3.1. SIP Config ............................................................................................................ 29
4.3.3.2. IAX2 Config ......................................................................................................... 33
4.3.3.3. Stun Config ......................................................................................................... 34
4.3.3.4. DIAL PEER setting ............................................................................................... 35
4.3.4. Phone ........................................................................................................................ 38
4.3.4.1. DSP Config .......................................................................................................... 38
4.3.4.2. Call Service ......................................................................................................... 39
4.3.4.3. Digital Map Configuration .................................................................................. 40
4.3.4.4. Phone Book ........................................................................................................ 42
4.3.5. Maintenance ............................................................................................................. 42
4.3.5.1. Auto Provision .................................................................................................... 42
4.3.5.2. Syslog Config....................................................................................................... 43
4.3.5.3. Config Setting ..................................................................................................... 44
4.3.5.4. Update ................................................................................................................ 45
4.3.5.5. Account Config ................................................................................................... 45
4.3.5.6. Reboot ................................................................................................................ 46
4.3.6. Security ..................................................................................................................... 47
4.3.6.1. MMI Filter ........................................................................................................... 47
4.3.6.2. Firewall ............................................................................................................... 47
4.3.6.3. NAT Config .......................................................................................................... 49
4.3.6.4. VPN Config .......................................................................................................... 52
4.3.7. Logout ........................................................................................................................ 53
4.4. Settings via phone’s keyboard. ....................................................................................... 54
4.4.1. How to set via the phone’s keyboard. ....................................................................... 54
4.4.2. Phone menu ............................................................................................................... 54
5. Appendix .................................................................................................................................54
5.1. Specification .................................................................................................................... 54
5.1.1. Device specification ................................................................................................... 54
5.1.2. Voice Features ........................................................................................................... 55
5.1.3. Network Features ...................................................................................................... 55
5.1.4. Maintenance and Management ................................................................................ 55
5.2. Digit-character map table ............................................................................................... 56
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
4
SVP309 Home SIP Phone User Manual
1. Introducing SVP309 VoIP Phone
1.1. Thank you for your purchasing SVP309
Thank you for your purchasing SVP309, SVP309 is a full-feature telephone that provides voice communication over the
same data network that your computer uses. This phone functions not only much like a traditional phone, allowing to
place and receive calls, and enjoy other features that traditional phone has, but also it own many data services features
which you could not expect from a traditional telephone.
This guide will help you easily use the various features and services available on your phone.
1.2. Delivery Content
Please check whether the delivery contains the following parts:
The base unit with display and keypad
The handset
The handset cable
The power supply
The Ethernet cable
1.3. Keypad
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
5
SVP309
9 Home SIP Phone User Manual
The numeric keypad with the keys 0 to 9, *, and # is used to enter
Digits and letters, additionally, the following keys are available:
Key mapping:
Key
Key name
Menu
Phone Book
Callers
LED
System Information
Confirm
Exit
Navigation Key
Function Description
In idle state
,press the MENU key to call up the menu.
In idle mode, press the Phone Book key to check the record list and
add new records and revise the record. Press this key again will
return to idle mode.
In idle/pickup/calling mode, press the Callers key to Check the
Income/Outgoing/Missed calls records. Press this key
ke again will
return to idle mode
LED blinks to remind user new voicemail.
In idle mode, press the Sysinfo key to check the phone setting
parameters. Such as local phone number, local IP and local Gateway
IP address.
Use the Enter key to enter next menu, or confirm the setting.
Use the Exit key to return to previous menu, ,cancel the setting, or
reject to answer a call.
When you pick up the handset or during calling, you can use this key
to turn up or turn down the handset volume; when a call comes, you
can use this key to adjust ring volume; you also can use this key to
choose item in the menu, callers or phone book.
MWI
Use this key to read old or new message.
Transfer
Use the key to realize blind transfer or attended transfer please
refer to 3.1.4.-call
call transfer for more details).
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
6
SVP309
9 Home SIP Phone User Manual
Conference
Delete
Hold
mute
Use this key to realize the three party call ( pleae refer
to3.1.5-Calling Hold and 3 ways call for more details)
In menu, use this key to modify current setting or delete invalid
information. And when you input number do call out, you can use
it to delete characters
Temporarily hold the active call during the talking; press the key
again to unhold the call. You also can press this key then input the
third party’s phone number and end with the # key during calling,
you can make a call with the third party and hold the previous
calling. (3.1.5-Calling
Calling Hold and 3 ways call).
Press this key in calling mode, you can hear the other side, and the
other side can not hear you
R/send
In the hook off /hands-free
free mode, use the key to dial the last call
number; use this key to make a quick dial as soon as you select your
desired number in phone book or callers.
Handfree
Enter into hands-free mode.
1.4. Port for connecting
POWER
DC
Power switch
Power port
Select ON/OFF
Output: 5V/1.0A
LAN
Network port
Connect it to PC
WAN
Network port
Connect it to Network
The phone has two Network ports: The WAN port and the LAN port. Before you connect the power, please carefully read
Safety Notices of this user manual.
2.Initial connecting and Setting
2.1. connect the phone
Step 1: Connect the IP Phone to the corporate IP telephony network. Before you connect the phone to the network,
please check if your network can work normally.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
7
SVP309 Home SIP Phone User Manual
You can do this in one of two ways, depending on how your workspace is set up.
Direct network connection—by this method, you need at least one available Ethernet port in your workspace. Use the
Ethernet cable in the package to connect WAN port on the back of your phone to the Ethernet port in your workspace.
Since this VoIP Phone has router functionality, whether you have a broadband router or not, you can make direct network
connect. The following two figures are for your reference.
Shared network connection—Use this method if you have a single Ethernet port in your workspace with your desktop
computer already connected to it. First, disconnect the Ethernet cable from the computer and attach it to the WAN port
on the back of your phone. Next, use the Ethernet cable in the package to connect LAN port on the back of your phone to
your desktop computer. Your IP Phone now shares a network connection with your computer. The following figure is for
your reference.
Step 2:
Connect the handset to the handset port by the handset cable in the package.
Step 3: connect the power supply plug to the DC port on the back of the phone. Use the power cable to connect the
power supply to a standard power outlet in your workspace.
Step 4: push the on/off switch on the back of the phone to the on side, then the phone’s LCD screen displays “WAIT
LOGON”. Later, a ready screen typically displays the date, time and current network mode.
If your LCD screen displays different information from the above, you need refer to the next section “Initial setting” to set
your network online mode.
If your VoIP phone registers into corporate IP telephony Server, your phone is ready to use.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
8
SVP309
9 Home SIP Phone User Manual
2.2. Initial Setting
This VoIP Phone provides you with rich function and parameters setting. If you have enough knowledge about network
and SIP protocol, it is better
er for you to understand many parameters. But if you know little about network and SIP
protocol, you can also easily make initial setting according to the following steps to enjoy rapidly high quality voice and
low cost from this VoIP Phone.
nitial setting, please check if your corporate IP telephony network can work normally, and you have finished
Before make initial
“connect the phone”.
This VoIP Phone Supports DHCP by default. It will receive an IP address and other network-related
network
settings (Netmask, IP
gateway,
ay, DNS server) from the DHCP server. If your network supports DHCP, you can connect this VoIP Phone directly to
the network. If your network doesn’t support DHCP, you need change this VoIP Phone’s network connection setting.
According to the following steps,
ps, change this VoIP Phone’s DHCP network connection setting into PPPoE or static IP which
your network supports at present.
2.2.1. PPPoE mode.
1. Press the 3 key for three seconds, then confirm it by the Enter key, your phone network connection mode will switch
into PPPoE mode. Prepare your PPPoE account name and password.
2. Press the MENU key, the LCD screen will display “INPUT PASSWORD”.
3. Input the password (default value is 123), and
and press the ENTER key, the LCD screen will display “NETWORK”.
4. Press the
key and LCD screen will display “LAN”, press the
screen will display “STATIC NET”. Then press the
display “USER NAME”.
5. Press the
key and then press the
key, enter it by the
key again, enter it by the
key, the LCD
key, the LCD screen will
key, input your PPPOE account number then press the
key to confirm. The LCD screen will display the inputted PPPOE account number.
6. Press the
key to return to the
th previous menu, then press the
key, the LCD screen will display
“PASSWORD”. Then press the ENTER key, and the DEL key, input your PPPoE’s password and confirm it by the
Key, the LCD screen will display the password which you inputted.
7. Press the EXIT
IT key for four times and press the DOWN key,
key, till the LCD screen display “SYSTEM”.
8. Press the ENTER key, the screen display “SAVE”, then press the
YOU SURE”.
key again, the LCD screen will display “ARE
key, the phone will save your setting and the LCD screen will display “SAVING NOW”, then return
9. Press the
to display “SAVE”.
10. Press the
key twice, then press numeric key “3”and hold until the screen display “ARE YOU SURE”. Press
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
9
SVP309
9 Home SIP Phone User Manual
the
key, the screen will display
ay “CHANGING”,, which means that the phone is trying to switch to PPPoE mode.
If the icom “PPPoE” on the top of the screen keeps blink, it shows that the phone is trying to access the PPPoE server.,
and the IP is still static IP if you press
key to displayy the current IP; if the icon “PPPoE” is showed without
blink, it means that the phone has already gotten IP from PPPoE server.
2.2.2. Static IP mode:
1. Press the 1 key for three seconds, then confirm it by the
key, your phone network connection mode will
switch into Static IP mode. Prepare your phone’s network parameters. They are IP Address of this phone, Subnet Mask,
Default Gateway/ Router and DNS. You can ask your VoIP service provider for those parameters.
2. Press the
key, the LCD screen will display “INPUT PASSWORD”.
3. Input password (default is 123), then press the
key, the LCD screen will display” NETWORK”.
4. Press the
key, and the LCD screen will display “LAN”. Press the
screen will display “STATIC NET”.
5. Press the
key, then the
key, the LCD screen will display “IP”. Press the
key again and then the
your desired IP address for your IP phone and confirmed by pressing the
inputted IP address. When inputting IP with keypad, use “*” instead of “.”.
6. Press the
Press the
key to return to previous menu, then press the
key then the
key, the LCD
key, input
key, then the LCD will display the
key, the LCD screen will display “DNS2”.
key, input your
our spare DNS address and confirm it by pressing the
key, and then the LCD will display the inputted DNS address.
7. Press the
Press the
key to return to previous menu, then press the
key then the
key, the LCD screen will display “DNS”.
key, inputt your DNS address and confirm it by pressing the
key,
and then the LCD will display the inputted DNS address.
8. Press the
key to return to the previous menu, and then press the
“GATEWAY”. Press the
pressing the
9. Press the
key again and then
th the
key, the LCD screen will display
key, input your gateway’s IP address and confirm it by
key, the LCD screen will display the inputted gateway address.
key to return to the previous menu, and then press the
key, the LCD screen will display
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
10
SVP309
9 Home SIP Phone User Manual
“NETMASK”. Press the
key again and then the
key, input your netmask and press the
key
to confirm it. The LCD screen will display the inputted netmask.
10. Press the
key for four times and press the
key, till the LCD Screen displays “SYSTEM”.
11. Press the
key, the LCD screen will display “save”, then press the
display” ARE YOU SURE”.
key again, the LCD screen will
12. Press the
key, this phone will display “SAVING NOW”, then return to display “SAVE”.
13. Press the
key twice to exit the menu, and then press the numeric key 1 till the LCD screen displays “ARE
YOU SURE”. Press the
key, the LCD screen will display “CHANGING”. If the icon “static” on the top of screen
shows without blink, it means phone has already used
us the static IP.
2.2.3. DHCP mode
Press the numeric key 2 and hold till the LCD screen displays “ARE YOU SURE”. Press the
key, the LCD screen
will display “CHANGING” and this VoIP phone is trying to switch to DHCP mode. If the icom “DHCP” on the top of the
screen keeps blink, it shows that the phone is trying to access the DHCP server., and the IP is 0.0.0.0 if you press
key to display the current IP; if the icon “DHCP” is showed without blink, it means that the phone has already gotten IP
from DHCP server.
3. Basic Functions
3.1. Basic operation
3.1.1. Accepting a call
There are four methods to accept an incoming call:
Pick up handset to accept incoming calls.
Press the
If you need switch from a hands-free
free call to handset, please pick up the handset directly.
If you need switch from a handset call to hands-free,
hands
please press the
button
button, and then hang up the
handset.
3.1.2. Making a call
Quick-dialing
In idle mode, input
nput the called number, and press # key or
button, phone will diall the call and use hands-free
hands
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
11
SVP309
9 Home SIP Phone User Manual
automatically.
Use handset
Pick up the handset, and the LCD screen will display “PLEASE
“
DIAL”” and you will hear dialing tone at the same time, then
input the phone number and end by the # button. When you hear long ring “du,, du…” from handset and the LCD screen
display “CALLING”,
”, the call is through. Hang up the handset to end the call.
Use hands-free
button and the LCD screen will display “PLEASE
“
DIAL”” and you will hear dialing tone at the same time,
Press the
then input the phone number and end by the # button. When you hear long ring “du, du…” and the LCD screen display
“CALLING”, the call is through. Press the
button again to end the call.
Use the phone book
Press the
button then the
button you will enter into the phone book. Press the
button to select your desired contact person, then press the
button to dial the call.
Use Callers
Press the
the
/
key, then select your desired phone number in callers by the
/
key, and next press
button to dial the call.
Use the R/send key
Please pick up or press the
key. After you hear dialing tone, please press the
key to dial the last
phone number. Note: after you reboot the phone, the phone will delete callers and Redial will be
b invalid.
3.1.3. Ending a call
Hangs up by handset onhook
Hangs up by press
Hangs up a call in call waiting state.
when in hands-free
hands
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
12
SVP309
9 Home SIP Phone User Manual
If you are in call waiting state, you could press # key to hang up the current call, and switch to the other call to keep
talking.Note: Pressing # key will not hang up if there is only one call currently.
3.1.4. Transferring a call
Call transfer has several ways to realize:
1. When A talks to B, B may press the
presses the
key and dial C phone number. After B talks to C ( or B hear alert from C ), B
key, then B hangs up, and A will get through to C.
2. When A is talking with B, C calls B, B may press the
key to hold A, and talk to C. Then B presses the
key,
A will get through to C.
3. When A talks to B, B presses the
key, dial C phone number and # key, then hang up and A will get through to
C.
1 and 2 are attended transfer; 3 is blind transfer.
Notice to VoIP Phone Carrier: Your VoIP phone server need support FRC3515, or else transferring can not work.
3.1.5. Calling Hold and 3 ways call
There are two modes to enjoy hold function:
1. Press the
key during a call, and the call will be on hold. While a call is on hold, you can establish another call by
dialing your desired number and confirm it by the # button. Pressing the
key again you will resume the first call.
By using hold function, you can talk with only one party; the other party who is on hold can’t talk with you. If you press
the * button or
key you will enter into 3 ways call.
2. If the third party calls you during a call, the LCD screen will display the incoming call number. Press the
key to
hold the first call, and then you can talk with the third party. By using hold function, you can talk with only one party;
the other party who is on hold can’t talk with you. If you press # key, phone will hang up the first call, and then accept
the new incoming call.
Notice:: You must enable the calling waiting or else calling hold can’t work.
3.1.6. Callers
The VoIP phone maintains lists of missed, received, and dialed calls. Each list can contain up to 100 entries. If the call
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
13
SVP309
9 Home SIP Phone User Manual
list capacity is full,
ll, new call will replace the first call. If you stop power supply or restart the phone, the record will
disappear.
Missed Calls
Press the
key, and then the
/
key, till the LCD screen display “MISSED”.
“
Press the
key, the LCD screen will display the missed call number and sequence numbersof the missed call.
You can press the
key to dial this phone number, you also press UP/DOWN key to browse the other missed
calls or you can press the
key again, the LCD screen
screen will display the time of the missed calls. If there is no
one missed calls, the LCD will display “LIST
LIST IS EMPTY”.
EMPTY
Received Calls
Press the
key, and then the
/
key, till the LCD screen display “RECEIVED”.
“
Press ENTER key,
the LCD screen will display the received call number and sequence number of the received call.
You can press the
key to dial this phone number, you also press
/
key to browse the other
received calls or you can press the
key again, the LCD screen will show the time of the received call. If
there is no one received call, the LCD will display “LIST
“
IS EMPTY”.
Dialed calls
Press the
key, and then the
/
key, till the LCD screen display "OUTGOING".
"
Press
key, the LCD screen will display the phone number and sequence number of the dialed call. You can press the
key to dial this phone number, or press the
/
key to browse all record of the dialed calls. If there is on
one dialed calls, the LCD will display “LIST
LIST IS EMPTY”.
EMPTY
3.2. The high-level operation
This VoIP Phone provides more advanced functions after setting at the permission scope of SIP server.
3.2.1. Special Keys
Realize Secondary Dial by Dialing for only one time
When you make secondary dial in off-hook/handsfree/standby
hook/handsfree/standby pre-input
pre
mode, press
key to postpone input, and
screen display will show--. One --stands
stands for 2 seconds. For example, you input 123--45,
123 45, the phone will send DTMF(45) 2
seconds after the phone call 123. 123------45
45 will make phone send DTMF(45) at 6 seconds interval.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
tele.com
14
SVP309 Home SIP Phone User Manual
MWI(Message Waiting Indication)
When a new voicemail coming, LED on the phone will flash. You can press the MWI key to listen new voicemail if you
configure mwi number
3.2.2. Call pickup
Call pickup is implemented by simulating pickup function of PBX. it’s that, when A calls B, B rings but no answer, at this
moment, C can hook off and input an appointed prefix plus B’s number, pick up A’s call and talk with A
The following chart shows how to configure an appointed prefix in dial peer to have call pick up function.
*1* means appointed prefix code. After making the above configuration, C can dial *1* plus B’phone number to pick up
A’s call. User can set prefix in random, in the case of no affecting current dialing rules.
3.2.3. join call
When B is calling C, A can join in the existing call by inputing an appointed prefix numbers plus B or C number, if B or C
also supports join call
The following chart shows how to configure an appointed prefix in dialpeer to have join call function.
*2* means appointed prefix code. After making the above configuration, A can dial *2* plus B or C number to join B and
C’s call, . User can set prefix in random, in the case of no affecting current dialing rules.
3.2.4. redial/unredial
If B is in busy line when A calls B, A will get notice: busy, please hang up. If A want to connect B as soon as B is in idle, he
can use redial function at the moment and he can dials an appointed prefix number plus B’s number to realize redial
function.
What is redial function? A can’t not build a call with B when B is in busy ,then A will subscribe B’s calling mode at 60
second intervals. once B is available, A will get reminder of rings to hook off, while A hooks off, A will call B automatically.
If at this time A is occupied temporarily and unwilling to contact B, A also can cancel the redial function by dialing an
appointed prefix plus B’s number before making the redial function.
*3* is appointed prefix code. After making the above configuration, A can dial
*3* plus B’phone number to make the redial function.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
15
SVP309 Home SIP Phone User Manual
*4* is appointed prefix code. After configuration, A can dial *4* to cancel redial function.
User can set prefix in random, in the case of no affecting current dialing rules.
3.2.5. click to dial
When user A browses in an appointed Web page, user A can click to call user B via a link (this link to user B), then user A’s
phone will ring, after A hooks off, the phone will dial to B.
4. Setting
4.1. Setting methods
VoIP Phone is different from the traditional phone; it need be set to make it active. If your VoIP service provider asks you
to set this phone, you can do it easily according to the following methods.
This VoIP Phone can be set via three different setting methods:
The phone key. The initial password is 123 for setting via phone key.
The web browser on PC
Telnet
This Manual will tell you about the setting methods via the web browser on PC.
4.2. Setting via Web Browse
When this phone and your PC are connected to your network, enter the IP address of the wan port in this phone as the
URL (e.g. http://xxx.xxx.xxx.xxx/ or http://xxx.xxx.xxx.xxx:xxxx/).
If you do not know the IP address, you can look it up on the phone’s display by pressing the key “SYSINFO”.
After you enter the IP address, you will see the following web interface.
This phone provides different two privileges for different users to set it.
The two privileges are guest and administrator respectively. In guest privilege, user can see but not modify Register/Proxy
Sever Addresses and ports of SIP, advance SIP and Iax2. In administrator privilege, user can see and modify all setting
parameters.
Default value in guest privilege
Username: guest
Password: guest
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
16
SVP309 Home SIP Phone User Manual
Default value in Administrator privilege
Username: admin
Password: admin
Input username and password, click “logon”, and you will enter setting web interface.
There is a selection menu on the left side of the web interface. Click on the desired submenu; the current settings of this
submenu will be displayed in the larger field on the right. You can now modify and store the values by using mouse and
keyboard of your PC. To save the changes, click on the submenu “maintenance” and then click the “ config” button and
the “Save” button on the right field.
4.3. Configuration via WEB
4.3.1. BASIC
4.3.1.1. Status
Status
Field name
Network
Phone Number
Explanation
Shows the configuration information on WAN and LAN port, including the
connect mode of WAN port (Static, DHCP, PPPoE), MAC address, the IP address
of WAN port and LAN port, ON or OFF of DHCP mode of LAN port.
Shows the phone numbers provided by the SIP LINE 1-2 servers.
The last line shows the version number and issued date.
4.3.1.2. Wizard
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
17
SVP309 Home SIP Phone User Manual
Wizard
Field Name
Explanation
Please select the proper network mode according to the network condition. SVP309 provide three different
network settings:
Static: If your ISP server provides you the static IP address, please select this mode, then finish Static
Mode setting. If you don’t know about parameters of Static Mode setting, please ask your ISP for them.
DHCP: In this mode, you will get the information from the DHCP server automatically; need not to input
this information artificially.
PPPoE: In this mode, your must input your ADSL account and password.
You can also refer to 3.2.1 Network setting to speed setting your network.
, 【
】
】
Choose Static IP MODE click NEXT can config the network and SIP(default SIP1)easily, also can browse
them too. Click BACK can return to the last page.
【
Static IP Address
Netmask
Gateway
Input the IP address distributed to you.
Input the Netmask distributed to you.
Input the Gateway address distributed to you.
DNS Domain
Set DNS domain postfix. When the domain which you inputted can not be
parsed, phone will automatically add this domain to the end of the domain
which you inputted before and parse it again.
Primary DNS
Input your primary DNS server address.
Alter DNS
Input your standby DNS server address.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
18
SVP309 Home SIP Phone User Manual
Display Name
Server Address
Server Port
If user set the display name, callee will show this display name.
Input your SIP server address.
Set your SIP server port.
User Name
Password
Phone Number
Input your SIP register account name.
Input your SIP register password.
Input the phone number assigned by your VOIP service provider.
Enable Register
Start to register or not by selecting it or not.
Display detailed information that you manual config.
Choose DHCP MODE click NEXT to config simple SIP(default SIP1). You can browse it too. Click BACK
to return to the last page. Like Static IP MODE
Choose PPPoE MODE click NEXT to config the PPPoE account/password and SIP(default SIP1). You can
browse it too. Click BACK to return to the last page. Like Static IP MODE
, 【 】
, 【 】
【 】
PPPoE Server
Username
Password
【
】
【
。
】
。
It will be provided by ISP.
Input your ADSL account.
Input your ADSL password.
Notice: Click Finish button after finish your setting, IP Phone will save the setting automatically and reboot.
After reboot, you can dial by the SIP account.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
19
SVP309 Home SIP Phone User Manual
4.3.1.3. Call Log
You can look up all the outgoing calls through this page.
Call Log
Field name
Start Time
Last Time
Called Number
explanation
Display the start time of the outgoing call
Display the conversation time of the outgoing call.
Display the account/protocol/line of the outgoing call.
4.3.1.4. MMI SET
MMI SET
Field name
Language Set
explanation
Set the language of phone, English is default.
4.3.2. Network
4.3.2.1. WAN Config
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
20
SVP309 Home SIP Phone User Manual
WAN Config
Field Name
Active IP
explanation
The current IP address of the phone.
Current Netmask
The current Netmask address.
MAC Address
Current Gateway
The current MAC address of the phone.
The current Gateway IP address.
Get MAC Time
Shows the time of getting MAC address
Please select the proper network mode according to the network condition. SVP309 provide three different
network settings:
Static: If your ISP server provides you the static IP address, please select this mode, then finish Static Mode
setting. If you don’t know about parameters of Static Mode setting, please ask your ISP for them.
DHCP: In this mode, you will get the information from the DHCP server automatically; need not to input this
information artificially.
PPPoE: In this mode, your must input your ADSL account and password.
You can also refer to 3.2.1 Network setting to speed setting your network.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
21
SVP309 Home SIP Phone User Manual
If you use static mode, you need set it.
IP Address
Input the IP address distributed to you.
Netmask
Gateway
DNS Domain
Primary DNS
Alter DNS
If you uses PPPoE mode
PPPoE Server
Username
Password
Input the Netmask distributed to you.
Input the Gateway address distributed to you.
Set DNS domain postfix. When the domain which you inputted can not be
parsed, phone will automatically add this domain to the end of the domain
which you inputted before and parse it again.
Input your primary DNS server address.
Input your standby DNS server address.
, you need to make the above setting.
It will be provided by ISP.
Input your ADSL account.
Input your ADSL password.
Notice:
1 Click “Apply” button after finishe your setting, IP Phone will save the setting automatically and new setting will
take effect.
2 If you modify IP address, the web will not response by the old IP address. Your need input new IP address in
the address column to logon in the phone.
3 If networks ID which is distributed by DHCP server is same as network ID which is used by LAN of system,
phone will use the DHCP IP to set WAN, and modify LAN’s networks ID(for example, system will change LAN IP
from 192.168.10.1 to 192.168.11.1) when phone uses DHCP client to get IP in startup; if phone uses DHCP
client to get IP in running status and network ID is also same as LAN’s, phone will refuse to accept the IP to
configure WAN.
)
)
)
4.3.2.2. LAN Config
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
22
SVP309 Home SIP Phone User Manual
LAN Config
Field name
explanation
LAN IP
Specify LAN static IP.
Netmask
Specify LAN Netmask.
DHCP Service
NAT
Select the DHCP server of LAN port or not. After user modify the LAN IP
address, phone will amend and adjust the DHCP Lease Table and save the
result amended automatically according to the IP address and Netmask. You
need restart the phone and the DHCP server setting will take effect.
Select NAT or not.
Select Bridge Mode or not: If you select Bridge Mode, the phone will no
Bridge Mode
longer set IP address for LAN physical port LAN and WAN will join in the same
network.. Click “Apply”, the phone will reboot.
Notice: If you choose the bridge mode, the LAN configuration will be disabled.
,
4.3.2.3. Qos Config
The VOIP phone support 802.1Q/P protocol and DiffServ configuration. VLAN functionality can use different VLAN IDs by
setting signal/voice VLAN and data VLAN. The VLAN application of this phone is very flexible.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
23
SVP309 Home SIP Phone User Manual
In chart 1, there is a layer 2 switch without setting VLAN. Any broadcast frame will be transmitted to the other ports
except the send port. For example, a broadcast information is sent out from port 1 then transmitted to port 2,3 and 4.
In chart 2, red and blue indicate two different VLANs in the switch, and port 1 and port 2 belong to red VLAN, port 3 and
port 4 belong to blue VLAN. If a broadcast frame is sent out from port 1, switch will transmit it to port 2, the other port in
the red VLAN and not transmit it to port3 and port 4 in blue VLAN. By this means, VLAN divide the broadcast domain via
restricting the range of broadcast frame transmition.
Note: chart 2 use red and blue to identify the different VLAN, but in practice, VLAN uses different VLAN IDs to identify.
QoS Configuration
Field name
VLAN Enable
VLAN ID Check Enable
explanation
Before select it to enable VLAN, you need enable Bridge mode in LAN config.
Enable VLAN ID check by selecting it. After enable VLAN ID check, if VLAN ID of a
data package is not the same with the phone’s or a data package do not have
VLAN ID, the data package will be discarded.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
24
SVP309 Home SIP Phone User Manual
Voice/Data VLAN
differentiated
DiffServ Enable
DiffServ Value
Voice 802.1P Priority
Data 802.1P Priority
After enable VLAN, system will set packets with different type of VLAN ID.
Undifferentiated means after using VLAN, both VoIP packets and other data
packets will use the voice VLAN ID; tag differentiated means after using VLAN,
VoIP(signal and voice) packets will add voice VLAN ID, and other data packets will
add data VLAN ID; data untaged means after using VLAN, only VoIP packets will
add voice VLAN ID. Other data packets will not use VLAN.
Select it or not to Enable or disable DiffServ.
Set DiffServ value, the common value is 0x00.
Specify 802.1P Priority of voice/signal data package.
Set 802.1p of data VLAN. Non-VoIP data (such as http, telnet, ping etc) will use
this value to set VLAN package.
Voice VLAN ID
Set VLAN ID of voice/signal data package.
Data VLAN ID
Set 802.1q of data VLAN ID. Non-VoIP data (such as http, telnet, ping etc) will use
this value to set VLAN package.
:
NOTICE
1 Startup VLAN, if set Voice/Data VLAN differentiated as Undifferentiated, all packets will use the Voice VLAN ID
as the tag.
2) Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and disable the DiffServ, then system
will not distinguish the voice and data, all packets will use the Voice VLAN ID as the tag.
3) Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and enable the DiffServ, then system
will distinguish the voice and data and add the VLAN ID each other.
4) Startup VLAN, if set Voice/Data VLAN differentiated as data untaged, then the packet of the signal/voice will
use the Voice VLAN ID as the tag, but the data packets will not take the VLAN tag.
5) If Disable the VLAN, regardless to set the Voice/Data VLAN differentiated or not, all packets will not take the
VLAN tag; If enable the DiffServ, all packets will only take the DiffServ value.
6) user need notice, enable the VLAN ID Check Enable that is default, If enable it, the phone will match the VLAN
ID strictly. When others' VLAN ID dismatch with us, the packets will discard. Contrarily, the phone will accept
the packets with the distinct VLAN ID.
7) You must gain the IP with the Static mode when you set VLAN, otherwise can't gain the IP in the VLAN and also
can not dial with point to point.
)
4.3.2.4. Service Port
You can set the port of telnet/HTTP/RTP by this page.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
25
SVP309 Home SIP Phone User Manual
SERVICE PORT
Field name
HTTP Port
Telnet Port
RTP Initial Port
explanation
,
set web browse port, the default is 80 port if you want to enhance system
safety you'd better change it into non-80 standard port
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
,
;
Set Telnet Port, the default is 23. You can change the value into others.
Example:
The IP address is 192.168.1.70. the telnet port value is 8023, the accessing
address is telnet 192.168.1.70 8023
Set the RTP Initial Port. It is dynamic allocation.
RTP Port Quantity
Set the maximum quantity of RTP Port, the default is 200.
Notice:
1 You need save the configuration and reboot the phone after set this page.
2 If you modify the port of Telnet and HTTP, you would better set the value more than 1024 because the port
value less than 1024 is system port reserved.
3 if you set 0 for the HTTP port, it will disable HTTP service.
)
)
)
4.3.2.5. DHCP SERVER
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
26
SVP309 Home SIP Phone User Manual
DHCP SERVER
Field name
DHCP Leased Table
Shows the DHCP Lease Table
Lease Table Name
Start IP
End IP
Netmask
Gateway
Lease Time
DNS
explanation
IP-MAC mapping table. If the LAN port of the phone connects to a device, this
table will show the IP and MAC address of this device.
, the unit of Lease time is Minute.
Specify the name of the lease table
Set the start IP address of the lease table
Set the end IP address of the lease table, the network device connected to LAN
port will get IP address between Start IP and End IP by DHCP.
Set the Netmask of the lease table
Set the Gateway of the lease table
Set the Lease Time of the lease table
Set the default DNS server IP of the lease table; Click the Add button to submit
and add this lease table
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
27
SVP309 Home SIP Phone User Manual
Select name of lease table, click the Delete button will delete the selected lease table from DHCP lease table.
DNS Relay
Select DNS Relay, the default is enable. Click the Apply button to become
effective.
Notice:
1 The size of lease table can not be larger than the quantity of C network IP address. We recommend you to use
the default lease table and not modify it.
2 If you modifies the DHCP lease table, you need save the configuration and reboot.
)
)
4.3.2.6. SNTP
Setting time zone and SNTP (Simple Network Time Protocol) server according to your location, you can also manually
adjust date and time in this web page.
SNTP
Field name
Server
Time Zone
Time Out
explanation
Set SNTP Server IP address.
Select the Time zone according to your location.
Set the time out, the default is 60 seconds.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
28
SVP309 Home SIP Phone User Manual
12 Hours Systems
SNTP
Swich the time mechanism between 12 hours and 24 hours.
Default is 24 hours mode
Select the SNTP, and click Apply to make the SNTP Times effective.
Enable Daylight
Enable daylight saving time
Time shift(minutes)
Setup the variety length
Month
Setup stat and end month
Week
Day
Setup start and end week
Setup start and end day
Hour
Setup start and end hours
Minute
Setup start and end minutes
Notice: You need specify the above all items.
4.3.3. VOIP
4.3.3.1. SIP Config
Set your SIP server in the following interface.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
29
SVP309 Home SIP Phone User Manual
SIP Config
Field name
explanation
【 】
Choose line to set info about SIP, there are 2 lines to choose. You can switch by
Load
button.
Register Status
Shows if the phone has been registered the SIP server or not; or so, show
Unapplied;
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
30
SVP309 Home SIP Phone User Manual
Server Name
Server Address
Server Port
Set the server name.
Input your SIP server address.
Set your SIP server port.
Account Name
Password
Input your SIP register account name.
Input your SIP register password.
Phone Number
Input the phone number assigned by your VoIP service provider. Phone will not
register if there is no phone number configured.
Display Name
Set the display name.
Proxy Server Address
Proxy Server Port
Set proxy server IP address Usually, Register SIP Server configuration is the same
as Proxy SIP Server. But if your VoIP service provider give different configurations
between Register SIP Server and Proxy SIP Server, you need make different
settings.
Set your Proxy SIP server port.
Proxy Username
Proxy Password
Input your Proxy SIP server account.
Input your Proxy SIP server password.
Domain Realm
Enable Register
(
)
Set the sip domain if needed, otherwise this VoIP phone will use the Register
server address as sip domain automatically. (Usually it is same with registered
server and proxy server IP address).
NAT Keep Alive Interval
Start to register or not by selecting it or not.
Set expire time of SIP server register, default is 60 seconds. If the register time of
the server requested is longer or shorter than the expire time set, the phone will
change automatically the time into the time recommended by the server, and
register again.
Set examining interval of the server, default is 60 seconds
User Agent
Signal Key
Media Key
Set the user agent if have, the default is VoIP Phone 1.0
Set the key for signal encryption
Set the key for RTP encryption
Local port
Set sip port of each line
Ring type
Hot line Number
Set ring type of each line
Set Hot line number of each line.
Transfer Expire Time
Enable Subscribe
The phone send bye and end the call as soon as hang up.
Enable Subscribe.
Enable Keep Authentication
Enable/Disable Keep Authentication.
Register Expire Time
NAT Keep Alive
Enable Via rport
Enable/Disable keeps NAT of SIP alive.
If some server refuse to register with too short interval time, and has no packets
sending to device in private network to keep NAT alive, user could set this
function ON. It need set the keep alive interval time less than the NAT server’s.
Enable PRACK
Enable/Disable system to support RFC3581. Via rport is special way to realize SIP
NAT.
Enable or disable SIP PRACK function, suggest use the default config.
Long Contact
Enable URI Convert
Set more parameters in contact field; connection with SEM server
Convert # to %23 when send the URI.
Dial Without Register
Set call out by proxy without registration;
Ban Anonymous Call
Set to ban Anonymous Call;
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
31
SVP309 Home SIP Phone User Manual
Forward Type
Select call forward mode, the default is Off
Off Close down calling forward
Busy If the phone is busy, incoming calls will be forwarded to the appointed
phone.
No answer If there is no answer, incoming calls will be forwarded to the
appointed phone.
Always Incoming calls will be forwarded to the appoint phone directly.
The phone will Prompt the incoming while doing forward.
:
:
:
:
Forward Phone Number
Appoint your forward phone number.
Server Type
Select the special type of server which is encrypted, or has some unique
requirements or call flows.
DTMF Mode
RFC Protocol Edition
Select DTMF sending mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Different VoIP Service providers may provide different modes.
Select SIP protocol version to adapt for the SIP server which uses the same
version as you select. For example, if the server is CISCO5300, you need to
change to RFC2543, else phone may not cancel call normally. System uses
RFC3261 as default.
Transport Protocol
Set transport protocols, TCP or UDP;
RFC Privacy Edition
Set Anonymous call out safely; Support RFC3323and RFC3325;
Subscribe Expire Time
Set the interval of Subscribe.
Enable DNS SRV
Support DNS looking up with _sip.udp mode
Click to Talk
Set click to Talk ( need practical software support).
Signal Encode
Enable/Disable Signal Encrypt.
RTP Encode
Enable/Disable RTP Encrypt.
Enable Session Timer
Set Enable/Disable Session Timer, whether support RFC4028.It will refresh the SIP
sessions.
Answer With Single Codec
Enable/Disable the function when call is incoming, phone replies SIP message
with just one codec which phone supports.
Auto TCP
Set to use automatically TCP protocol to guarantee usability of transport as
message is above 1300 byte
Enable Strict Proxy
Support the special SIP server-when phone recieves the patckets sent from
server phone will use the source IP address, not the address in via field.
Enable GRUU
Set to support GRUU
Enable Displayname Quote
Set to make quotation mark to displayname as the phone sends out signal, in
order to be compatible with server.
,
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
32
SVP309 Home SIP Phone User Manual
4.3.3.2. IAX2 Config
IAX2 Config
Field name
Register Status
IAX2 Server Addr
explanation
Shows if the phone has been registered the IAX2 server or not.
Input your IAX2 server address.
IAX2 Server Port
Set your IAX2 server port, the default is 4569.
Account Name
Account Password
Input your IAX2 register account name.
Input your IAX2 register password.
Phone Number
Input your assigned phone number (usually it is same you’re your IAX2 account
name).
Local Port
Voice Mail Number
Set your local sport the default is 4569.
Specify the voice mail’s number.
Voice Mail Text
Specify the voice mail’s name.
Echo Test Number
,
Set echo test number. If IAX2 server supports echo test, and echo test number is
non- numeric, system could set an echo test number to replace the echo test text. So
user can dial the numeric number to test echo voice test. This function is provided
with server to make endpoint to test whether endpoint could talk through server
normally.
Echo Test Text
Specify echo test text’s name.
Refresh Time
Set expire time of IAX2 server register, you can set it between 60 and 3600 seconds.
Enable Register
Start to register the IAX2 server or not by selecting it or not.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
33
SVP309 Home SIP Phone User Manual
Enable G.729
IAX2(Default Protocol)
Enable or disable code G.729 by selecting it or not
Enable or disable IAX2 as default dial protocol
4.3.3.3. Stun Config
In this web page, you can config SIP STUN.
STUN:
By STUN server, the phone in private network could know the type of NAT and the NAT mapping IP and port of SIP.
The phone might register itself to SIP server with global IP and port to realize the device both calling and being called in
private network.
STUN
Field name
STUN NAT Transverse
STUN Server Addr
explanation
Shows STUN NAT Transverse estimation, true means STUN can penetrate NAT, while False
means not.
Set your SIP STUN Server IP address
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
34
SVP309 Home SIP Phone User Manual
STUN Server Port
STUN Effect Time
Local SIP Port
Set your SIP STUN Server Port
Set STUN Effective Time. If NAT server finds that a NAT mapping is idle after time out, it will
release the mapping and the system need send a STUN packet to keep the mapping effective
and alive.
Set the SIP port.
Choose line to set info about SIP, There are 2 lines to choose. You can switch by
Use Stun
Enable/Disable SIP STUN.
【Load】 button.
Notice: SIP STUN is used to realize SIP penetration to NAT. If your phone configures STUN Server IP and Port (default is
3478), and enable SIP Stun, you can use the ordinary SIP Server to realize penetration to NAT.
4.3.3.4. DIAL PEER setting
This functionality offers you more flexible dial rule, you can refer to the following content to know how to use this dial rule.
When you want to dial an IP address, the entry of IP addresses is very cumbersome, but by this functionality, you can set
number 156 to replace 192.168.1.119 here.
When you want to dial a long distance call to Beijing, you need dial an area code 010 before local phone number, but you
can also dial number 1 instead of 010 after we make a setting according to this dial rule. For example, you want to dial
01062213123, but you need dial only 162213123 to realize your long distance call after you make this setting.
To save the memory and avoid abundant input of user,add the follow fuctions:
、
1 x Match any single digit that is dialed.
If user makes the above configuration, after user dials 11 digit numbers started with 13, the phone will send out 0 plus the
dialed numbers automatically.
、
2 [] Specifies a range that will match digit. It may be a range, a list of ranges separated by commas, or a list of digits.
If user makes the above configuration, after user dials 11 digit numbers started with from 135 to 139, the phone will send
out 0 plus the dialed numbers automatically.
Use this phone you can realize dialing out via different lines without switch in web interface.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
35
SVP309 Home SIP Phone User Manual
DIAL PEER
Field name
Phone number
explanation
There are two types of matching conditions: one is full matching, the other is
prefix matching. In the Full matching, you need input your desired phone number
in this blank, and then you need dial the phone number to realize calling to what
the phone number is mapped. In the prefix matching, you need input your
desired prefix number and T; then dial the prefix and a phone number to realize
calling to what your prefix number is mapped. The prefix number supports at
most 30 digits
Destination
Set Destination address. This is optional config item. If you want to set peer to
peer call, please input destination IP address or domain name. If you want to use
this dial rule in SIP2 line, you need input 255.255.255.255 or 0.0.0.2 in it.
Port
Alias
Set the Signal port, the default is 5060 for SIP.
Set alias. This is optional config item. If you don’t set Alias, it will show no alias.
Note: There are four types of aliases.
1) add: xxx, it means that you need dial xxx in front of phone number, which will reduce dialing number length.
2) all: xxx, it means that xxx will replace some phone number.
3) del: It means that phone will delete the number with length appointed.
4) Rep: It means that phone will replace the number with length and number appointed.
You can refer to the following examples of different alias application to know more how to use different aliases
and this dial rule.
Call Mode
Select differenct signal protocol, SIP or IAX2
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
36
SVP309 Home SIP Phone User Manual
Suffix
Set suffix, this is optional config item. It will show no suffix if you don’t set it.
Delete Length
Set delete length. This is optional config item. For example: if the delete length is
3, the phone will delete the first 3 digits then send out the rest digits. You can
refer to examples of different alias application to know how to set delete length.
Introduction of how to set up dial-peer to implement switch between multi- SIP lines
,
9T mapping: If you have registered a SIP1 server and set dial-peer according to the above table all calls will be
sent via SIP1 server when you press the numeric key “9” in front of dialing destination phone numbers.
8T mapping: If you have registered a Private SIP2 server and set dial-peer according to the above table all calls will
be sent via SIP2 server when you press the numeric key “8” in front of dialing destination phone numbers.
,
the rule of 2T means user need to dial the number with prefix 2 if he want to dial via IAX2 server
Examples of different alias application
Set by web
explanation
example
You need set phone number, Destination, Alias and If you dial “93333”, the
Delete Length.
SIP2 server will receive
Phone number is XXXT, Destination is “3333”
255.255.255.255 and Alias is del.
This means any phone No. that starts with your set
phone number will be sent via SIP2 line after the
first several digits of your dialed phone number are
deleted according to delete length.
This setting will realize speed dial function, after When you dial “2”, the SIP1
you dialing the numeric key “2”, the number after server
will
receive
all will be sent out.
33334444
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
37
SVP309 Home SIP Phone User Manual
The phone will automatically send out alias number When you dial “8309“, the
adding your dialed number, if your dialed number SIP1 server will receive
starts with your set phone number.
“07558309”
You need set Phone Number, Alias and Delete
Length. Phone number is XXXT and Alias is Rep:xxx When you dial “0106228”,
If your dialed phone number starts with your set the SIP1 server will receive
phone number, the first digits same as your set “86106228”
phone number will be replaced by the alias number
specified and New phone number will be send out.
If your dialed phone number starts with your set When you dial “147”, the
phone number. The phone will send out your dialed SIP1 server will receive
phone number adding suffix number.
“1470011”
4.3.4. Phone
4.3.4.1. DSP Config
In this page, you can configure voice codec, input/output volume and so on.
DSP Configuration
Field name
explanation
First Codec
The fist preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726
Second Codec
The second preferential DSP codec: G.711A/u, G.722, G.723, G.729,G.726
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
38
SVP309 Home SIP Phone User Manual
Third Codec
The third preferential DSP codec: G.711A/u, G.722, G.723, G.729,G.726
Forth Codec
The forth preferential DSP codec: G.711A/u, G.722, G.723, G.729,g.726
Fifth Codec
The fifth preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726
Sixth Codec
The sixth preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726
Input Volume
Handfree Volume
Specify Input (MIC) Volume grade.
Specify Handfree Volume grade
G729 Payload Length
Set G729 Payload Length
Handdown Time
Specify the least reflection time of Handdown, the default is 200ms.
Output Volume
Specify Output (receiver) Volume grade.
Ring Volume
Specify Ring Volume grade
G722 Timestamps
G723 Bit Rate
160/20ms or 320/20ms is available
5.3kb/s or 6.3kb/s is available
Default Ring Type
Set up the ring by default
Signal Standard
VAD
Select Signal Standard.
Select it or not to enable or disable VAD. If enable VAD, G729 Payload length
could not be set over 20ms.
;
4.3.4.2. Call Service
In this web page, you can configure Hotline, Call Transfer, Call Waiting, 3 Ways Call, Black List, white list Limit List and so
on.
Call Service
Field name
explanation
Hotline
Specify Hotline number. If you set the number, you can not dial any other numbers.
No Answer Time
Specify No Answer Time
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
39
SVP309 Home SIP Phone User Manual
P2P IP Prefix
Set Prefix in peer to peer IP call. For example: what you want to dial is
192.168.1.119, If you define P2P IP Prefix as 192.168.1., you dial only #119 to reach
192.168.1.119. Default is “.”. If there is no “.” Set, it means to disable dialing IP.
MWI Number
Set the number to listen record in server.
Do Not Disturb
Select NO Disturb, the phone will reject any incoming call, the callers will be
reminded by busy, but any outgoing call from the phone will work well.
Enable Call Transfer
Enable Call Transfer by selecting it.
Enable Call Waiting
Enable Call Waiting by selecting it.
Enable Three Way Call
Enable Three Way Call
Accept Any Call
If select it, the phone will accept the call even if the called number is not belong to
the phone.
Auto Answer
Ban Outgoing
If select it, the phone will auto answer when there is an incoming call.
If you select Ban Outgoing to enable it, and you can not dial out any number.
Black List
Set Add/Delete Black list. If user does not want to answer some phone calls, add
these phone numbers to the Black List, and these calls will be rejected.
x and . are wildcard. x means matching any single digit. for example, 4xxx expresses
any number with prefix 4 which length is 4 will be forbidden to dialed out
DOT (.) means matching any arbitrary number digit. for example, 6. expresses any
number with prefix 6 will be forbidden to dialed out.
if user wants to allow a number or a series of number incoming, he may add the
number(s) to the list as the white list rule. the configuration rule is -number, for
example, -123456, or -1234xx
Means any incoming number is forbidden except for 4119
Note: End with DOT (.) when set up the white list
Set Add/Delete Limit List. Please input the prefix of those phone numbers which you
Limit List
forbid the phone to dial out. For example, if you want to forbid those phones of 001
as prefix to be dialed out, you need input 001 in the blank of limit list, and then you
can not dial out any phone number whose prefix is 001.
x and . are wildcard. x means matching any single digit. for example, 4xxx expresses
any number with prefix 4 which length is 4 will be forbidden to dialed out
. means matching any arbitrary number digit. for example, 6. expresses any number
with prefix 6 will be forbidden to dialed out.
Notice: Black List and Limit List can record at most10 items respectively.
4.3.4.3. Digital Map Configuration
This phone supports 4 dial modes:
1). End with “#”: dial your desired number, and then press #.
2). Fixed Length: the phone will intersect the number according to your specified length.
3). Time Out: After you stop dialing and waiting time out, system will send the number collected.
4). User defined: you can customize digital map rules to make dialing more flexible. It is realized by defining the prefix of
phone number and number length of dialing.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
40
SVP309 Home SIP Phone User Manual
In order to keep some users' secondary dialing manner when dialing the external line with pbx, phone can be added a
special rule to realize it. so user can dial a number as external line prefix and get the secondary dial tone to keep dial the
external number. after finishing dialing, phone will send the prefix and external number totaly to ther server.
for example, there is a rule 9,xxxxxxxx in the digital map table. after dialing 9, phone will send the secondary dial tone,
user may keep going dialing. after finished, phone will call the number which starts with 9, actually the number sent out is
9-digit with 9.
Field name
End with "#"
Fixed Length
Digital Map Configuration
explanation
Set Enable/Disable the phone ended with “#” dial.
Specify the Fixed Length of phone ending with.
Set the timeout of the last dial digit. The call will be sent after timeout.
Time out
Below is user-defined digital map rule:
[] Specifies a range that will match digit. May be a range, a list of ranges separated by commas, or a list of digits.
x Match any single digit that is dialed.
. Match any arbitrary number of digits including none.
Tn Indicates an additional 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.
[1-8]xxx: Cause extensions 1000-8999 to be dialed immediately
9xxxxxxx: Cause 8 digit numbers started with 9 to be dialed immediately
911: Cause 911 to be dialed immediately after it is entered.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
41
SVP309 Home SIP Phone User Manual
99T4: Cause 99 to be dialed after 4 seconds.
9911x.T4:Cause any number started with 9911 to be dialed 4 seconds after dialing ceases.
Notice: End with “#”, Fixed Length, Time out and Digital Map Table can be used simultaneously, System will stop
dialing and send number according to your set rules.
4.3.4.4. Phone Book
You can input the name, phone number and select ring type for each name here.
Phone Book
Field name
explanation
Shows the detail of current phonebook.
Name
Number
Ring Type
Shows the name corresponding to the phone number
Shows the phone number
Shows the ring type of the incoming call.
Click “Modify” to change the selected information and click the “Delete” to delete the selected record.
Notice: the maximum capability of the phonebook is 500 items
4.3.5. Maintenance
4.3.5.1. Auto Provision
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
42
SVP309 Home SIP Phone User Manual
Auto Provision
Field name
Current Config Version
explanation
Show the current config file’s version.
Server Address
Set FTP/TFTP/HTTP server IP address for auto update. The address can be IP
address or Domain name with subdirectory.
Username
Set FTP server Username. System will use anonymous if username keep blank.
Password
Config File Name
Set FTP server Password.
Set configuration file’s name which need to update. System will use MAC as
config file name if config file name keep blank. For example, 000102030405.
Config Encrypt Key
Protocol Type
Input the Encrypt Key, if the configuration file is encrypted.
Update Interval Time
Update Mode
。
、
Select the Protocol type FTP TFTP or HTTP.
Set update interval time, unit is hour.
Different update modes:
1. Disable: means no update
2. Update after reboot: means update after reboot.
3. Update at time interval: means periodic update.
4.3.5.2. Syslog Config
Syslog is a protocol which is used to record the log messages with client/server mechanism. Syslog server receives the
messages from clients, and classifies them based on priority and type. Then these messages will be written into log by
some rules which administrator can configure. This is a better way for log management.
8 levels in debug information:
Level 0---emergency: This is highest default debug info level. You system can not work.
Level 1---alert: Your system has deadly problem.
Level 2---critical: Your system has serious problem.
Level 3---error: The error will affect your system working.
Level 4---warning: There are some potential dangers. But your system can work.
Level 5---notice: Your system works well in special condition, but you need to check its working environment and
parameter.
Level 6---info: the daily debugging info.
Level 7---debug: the lowest debug info. Professional debugging info from R&D person.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
43
SVP309 Home SIP Phone User Manual
At present, the lowest level of debug information send to Syslog is info, debug level only can be displayed on telnet.
Field name
Server IP
Syslog Configuration
explanation
Set Syslog server IP address.
Server Port
Set Syslog server port.
MGR Log Level
SIP Log Level
IAX2 Log Level
Set the level of MGR log.
Set the level of SIP log.
Set the level of IAX2 log.
Enable Syslog
Select it or not to enable or disable syslog.
4.3.5.3. Config Setting
Config Setting
Field name
Save Config
Backup Config
explanation
you can save all changes of configurations. Click the Save button, all
changes of configuration will be saved, and be effective immediately. .
Right clicks on “Right click here…” and select “Save Target As….” then you
will save the config file in .txt format
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
44
SVP309 Home SIP Phone User Manual
Clear Config
user can restore factory default configuration and reboot the phone.
If you login as Admin, the phone will reset all configurations and restore
factory default; if you login as Guest, the phone will reset all configurations
except for VoIP accounts (SIP1-2 and IAX2) and version number.
4.3.5.4. Update
You can update your configuration with your config file in this web page.
Update
Field name
Web Update
Server
Username
Password
File name
explanation
Click the browse button, find out the config file saved before or provided by
manufacturer, download it to the phone directly, press “Update” to save. You
can also update downloaded update file, logo picture, ring, mmiset file by web.
Set the FTP/TFTP server address for download/upload. The address can be IP
address or Domain name with subdirectory.
Set the FTP server Username for download/upload.
Set the FTP server password for download/upload.
Set the name of update file or config file. The default name is the MAC of the
phone, such as 000102030405.
Notice: You can modify the exported config file. And you can also download config file which includes several
modules that need to be imported. For example, you can download a config file just keep with SIP module. After
reboot, other modules of system still use previous setting and are not lost.
:
Type
Action type that system want to execute
1. Application update: download system update file
2. Config file export: Upload the config file to FTP/TFTP server, name and save it.
3. Config fie import: Download the config file to phone from FTP/TFTP server.
The configuration will be effective after the phone is reset.
Protocol
Select FTP/TFTP server
4.3.5.5. Account Config
You can add or delete user account, and change the authority of each user account in this web page
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
45
SVP309 Home SIP Phone User Manual
Field name
Keyboard Password
Account Configuration
explanation
Set the password for entering the setting menu of the phone by the phone ‘s
key board. The password is digit.
This table shows the current user existed.
User Name
Set account user name.
User Level
Set user level, Root user has the right to modify configuration, General can
only read.
Password
Set the password.
Confirm
Confirm the password.
Select the account and click the Modify to modify the selected account, and click the Delete to delete the
selected account.
General user only can add the user whose level is General.
4.3.5.6. Reboot
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
46
SVP309 Home SIP Phone User Manual
If you modified some configurations which need the phone’s reboot to be effective, you need click the Reboot, then the
phone will reboot immediately.
Notice: Before reboot, you need confirm that you have saved all configurations..
4.3.6. Security
4.3.6.1. MMI Filter
MMI Filter
User could make some device own IP, which is pre-specified, access to the MMI of the phone to config and
manage the phone.
Field name
explanation
MMI Fileter IP Table list:
Add or delete the IP address segments that access to the phone.
Set initial IP address in the Start IP column, Set end IP address in the End IP column, and click Add to add
this IP segment. You can also click Delete to delete the selected IP segment.
Select it or not to enable or disable MMI Filter. Click Apply to make it
effective.
Notice: Do not set your visiting IP outside the MMI filter range, otherwise, you can not logon through the
web.
MMI Filter
4.3.6.2. Firewall
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
47
SVP309 Home SIP Phone User Manual
Firewall Configuration
In this web interface, you can set up firewall to prevent unauthorized Internet users from accessing private
networks connected to the Internet (input rule), or prevent unauthorized private network devices from accessing
the Internet (output rule).
Firewall supports two types of rules: input_access rule and output_access rule. Each type supports at most 10
items.
Through this web page, you could set up and enable/disable firewall with input/output rules. System could
prevent unauthorized access, or access other networks set in rules for security. Firewall, is also called access list,
is a simple implementation of a Cisco-like access list (firewall). It supports two access lists: one for filtering input
packets, and the other for filtering output packets. Each kind of list could be added 10 items.
We will give you an instance for your reference.
Field name
explanation
In_access enable
Select it to Enable in_ access rule
out_access enable
Input/Output
Select it to Enable out_ access rule
Specify current adding rule by selecting input rule or output rule.
Deny/Permit
Specify current adding rule by selecting Deny rule or Permit rule.
Protocol Type
Filter protocol type. You can select TCP, UDP, ICMP, or IP.
Port Range
Set the filter Port range
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
48
SVP309 Home SIP Phone User Manual
Src Addr
Des Addr
Src Mask
Set source address. It can be single IP address, network address, complete
address 0.0.0.0, or network address similar to *.*.*.0
Set the destination address. It can be IP address, network address, complete
address 0.0.0.0, or network address similar to *.*.*.*
Set the source address’ mask. For example, 255.255.255.255 means just point to
one host; 255.255.255.0 means point to a network which network ID is C type.
Set the destination address’ mask. For example, 255.255.255.255 means just
point to one host; 255.255.255.0 means point to a network which network ID is
C type.
Click the Add button if you want to add a new output rule.
Des Mask
Then enable out_access, and click the Apply button.
So when devices execute to ping 192.168.1.118, system will deny the request to send icmp request to
192.168.1.118 for the out_access rule. But if devices ping other devices which network ID is 192.168.1.0, it will
be normal.
Click the Delete button to delete the selected rule.
4.3.6.3. NAT Config
NAT is abbreviated from Net Address Translation; it’s a protocol responsible for IP address translation. In other word, it is
responsible for transforming IP and port of private network to public, also is the IP address mapping which we usually say.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
49
SVP309 Home SIP Phone User Manual
:
DMZ config
In order to make some intranet equipments support better service for extranet, and make internal network security more
effectively, these equipments open to extranet need be separated from the other equipments not open to extranet by the
corresponding isolation method according to different demands. We can provide the different security level protection in
terms of the different resources by building a DMZ region which can provide the network level protection for the
equipments environment, reduce the risk which is caused by providing service to distrust customer, and is the best
position to put public information
The following chart describes the network access control of DMZ
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
50
SVP309 Home SIP Phone User Manual
NAT Configuration
Field name
IPSec ALG
FTP ALG
PPTP ALG
explanation
It is an encryption technology. Select it to enable IPSec ALG, the default is
enable
FTP is a service of connection layer which can transform intranet IP into
extranet IP when intranet IP is sending out packet.
Select it to enable FTP ALG, the default is enable
Select it enable PPTP ALG, the default is enable
Shows the NAT TCP mapping table
Shows the NAT UDP mapping table
Transfer Type
Inside IP
Select the NAT mapping protocol style, TCP or UDP
Set the IP address of device which is connected to LAN interface to do NAT
mapping.
Inside Port
Set the LAN port of the NAT mapping
Outside Port
Set the WAN port of the NAT mapping
Notice: After finish setting, click the Add button to add new mapping table; click the Delete button to delete
the selected mapping table.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
51
SVP309 Home SIP Phone User Manual
Shows the outside WAN port IP address and the inside LAN port IP address.
Outside IP
Set the outside Wan port IP address of DMZ.
Inside IP
Set the inside LAN port IP address of DMZ
Click the Add button to add new table; click the Delete button to delete the selected mapping table.
Notice: 10M/100M adaptive means the network card, and other equipment physical consultations speed,
testing speed under bridge mode near to 100M, in order to ensure the quality of voice and communications
real-time performance, we made some sacrifices of NAT under the transmission performance. Transmit with
full capability only when system is idle, so can not guarantee that the transmission speed reach to 100M.
4.3.6.4. VPN Config
This web page provides us a safe connect mode by which we can make remote access to enterprise inner network from
public network. That is to say, you can set it to connect public networks in different areas into inner network via a special
tunnel.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
52
SVP309 Home SIP Phone User Manual
VPN Configuration
Field name
VPN IP
explanation
Shows the current VPN IP address
Select UDP Tunnel (VPN Tunnel) or VPN L2TP. You can choose only one for current state. After you select it,
you’d better save configuration and reboot your phone.
Enable VPN
Select it or not to enable or disable VPN
;
VPN Server Addr
VPN Server Port
Set VPN Server IP Address
Set VPN Server Port
VPN Server Addr
VPN User Name
Set VPN L2TP Server IP address
Set User Name access to VPN L2TP Server
VPN Password
Set Password access to VPN L2TP Server
4.3.7. Logout
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
53
SVP309 Home SIP Phone User Manual
,
Click Logout and you will exit web page. If you want to enter it next time, you need input user name and password again.
4.4. Settings via phone’s keyboard.
4.4.1. How to set via the phone’s keyboard.
Press Menu, Up/Down, Enter and exit key to browse, select, and cancel
Use the Up/Down key to browse the menu and submenu
Use the ENTER key to enter into submenu and confirm your operation, the EXIT key can be used to back and cancel
operation.
4.4.2. Phone menu
Phone main menu:
5. Appendix
5.1. Specification
5.1.1. Device specification
Item
Adapter(Input/Output)
Port
:
~
this VoIP Phone
:
WAN
Input 100-240VAC 50 60Hz
Output 5V/1A
10/100Base- T RJ-45 for LAN, Auto MDIX
LAN
10/100Base- T
RJ-45 for PC, Auto MDIX
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
54
Power Consumption
LCD size
Operation Temperature
Relative Humidity
Main Chipset
Size
:
0~40℃
10~65%
275MHZ MIPS
16Mbits
Flash
2Mbits
Weight
:
Idle 1.5W/Active 1.8W
74 x 28mm
SDRAM
(W x H x D)
SVP309 Home SIP Phone User Manual
11.6×8×3 in.(295×205×75mm)
2.07lb.(0.94kg)
5.1.2. Voice Features
Support 2 lines SIP and IAX2, SIP 2.0 (RFC3261)
Codec G.711A/u G.7231 high/low G.729, G.722,G.726
Echo cancellation Support G.168 and hand-free can support 96ms
Support VAD CNG
NAT transverse: support STUN
Supports full duplex.
MD5
DNS name of server, peer to peer
SIP support SIP domain SIP authentication none basic
SIP support Pubic & Private server, user can through each server to calling in and out
DTMF SIP info DTMF Relay RFC2833
SIP application: contain SIP call forward/transfer/holding/waiting/3 way conference/Paging and intercom/ click to
dial/pickup/ joincall/redial/unredial.
Call control features: Flexible dial map, support hotline, empty calling no. reject server, black list for reject,
authenticated call, no disturb, caller ID and so on.
Support phonebook 500 records, incoming calls / outgoing calls / missing calls. Each supports 100 records
support conference call in server
Could dial use private server automatically when public server unregistered while private server is resgistered
successfully
Phonebook supports VCard standard
Support 12/24 time format.
12/24 hours time display
Support daylight saving time
Support path, gruu
Support SIP Privacy.
5.1.3. Network Features
WAN/LAN: support Bridge and Router mode.
Support basic NAT and NAPT
Support PPPoE for xDSL
support VLAN
Support NAT penetration, and Stun penetration
Support DMZ
Support VPN( L2TP. UDP)
Support DHCP get IP on WAN port
Support DHCP distribute IP on LAN port
Qos supports Diffserv.
support network tools: contain ping trace route telnet client
:
,
,
:
,
,
:
,
(
,
,
),
,
,
,
5.1.4. Maintenance and Management
The phone supports post mode, can update firmware by post mode.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
55
SVP309 Home SIP Phone User Manual
Supports different levels of administration.
Support Boot Monitor
Can upgrade firmware through boot monitor
access with different authority
support auto provisioning
Can config through Web, Keypad, Telnet
Can upgrade firmware and configuration file through HTTP, FTP, TFTP
Support syslog
5.2. Digit-character map table
Button
Character
1@
Button
Character
7PQRSpqrs
2ABCabc
8TUVtuv
3DEFdef
9WXYZwxyz
4GHIghi
.
5JKLjkl
0
6MNOmno
#
Copyright Notice:
No part of this document may be reproduced, transmitted, transcribed, stored in a retrieval system, or
translated into any language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical,
manual or otherwise without the prior written permission of Stephen Technologies Co.,Limited.
Disclaimer Notice
No license is granted, implied or otherwise, under any patent or patent rights of Stephen Technologies
Co.,Limited. Stephen Technologies Co.,Limited, makes no warranties, implied or otherwise, in regard to this
document and to the products described in this document. The information provided by this document is
believed to be accurate and reliable to the publication date of this document. However, Stephen Technologies
Co.,Limited assumes no responsibility for any errors in this document. Furthermore, Stephen Technologies
Co.,Limited, assumes no responsibility for the use or misuse of the information in this document and for any
patent infringements that may arise from the use of this document. The information and product specifications
within this document are subject to change at any time, without notice and without obligation to notify any
person of such change.
STEPHEN TECHNOLOGIES CO.,LIMITED/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115
Tel: +86 755 89352606 /Fax:+86 755 89352612 / Email: Support@stephen-tele.com / Url: www.stephen-tele.com
56
Download PDF
Similar pages