Cisco SVP2000P User Manual

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

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SVP2000P Business POE IP 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: [phone]

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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 behavi

or or induce 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, i

shock.

 The plug-

socket combination must be accessible at all times because it serves as the main disconnecting device.



Do not drop, knock or shake it. Rough handling can break internal circuit boards.

 Do not install the devi

ce 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 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 wit

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 place.

 You are in a situation that

could cause 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.

SVP2000PP

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℃

or high humidity. Avoid wetting the unit with any liquid.

POE IP Phone User Manual

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t may cause fire or electric

h a soft cloth that has been

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

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Table of Content

1.INTRODUCING SVP2000P VOIP PHONE .......................................................................... 5

1.1. T

HANK YOU FOR YOUR PURCHASING

SVP2000P ..................................................................... 5

1.2. D

ELIVERY CONTENT

.............................................................................................................. 5

1.3. K

EYPAD

.............................................................................................................................. 5

1.4. P

ORT FOR CONNECTING

........................................................................................................ 6

2. INITIAL CONNECTING AND SETTING .............................................................................. 7

2.1.

CONNECT THE PHONE

............................................................................................................ 7

2.1.1. Connect to network ................................................................................................. 7

2.1.2. Power adaptor connection ...................................................................................... 8

2.2. B

ASIC INITIALIZATION

........................................................................................................... 8

2.2.1. Network settings ...................................................................................................... 8

3. SVP2000P’S BASIC OPERATION ..................................................................................... 9

3.1. A

NSWER CALLS

.................................................................................................................... 9

3.2. P

LACE CALLS

......................................................................................................................10

3.3. E

ND CALLS

........................................................................................................................11

3.4. C

ALL TRANSFER

..................................................................................................................11

3.5. C

ALL HOLD

........................................................................................................................12

3.6. 3-

WAY CONFERENCE CALL

...................................................................................................12

3.7. S

WITCHBOARD OPERATOR FEATURE

.....................................................................................12

3.8. C

ALL RECORDS

...................................................................................................................12

4. WEB CONFIGURATION................................................................................................ 12

4.1. I

NTRODUCTION OF CONFIGURATION

.....................................................................................12

4.1.1. Ways to configure ..................................................................................................12

4.1.2. Password Configuration .........................................................................................13

4.2. S

ETTING VIA WEB BROWSER

................................................................................................13

4.3. C

ONFIGURATION VIA

WEB .................................................................................................13

4.3.1. BASIC ......................................................................................................................13

4.3.1.1. Status ...............................................................................................................13

4.3.1.2. Wizard .............................................................................................................14

4.3.1.3. Call Log.............................................................................................................16

4.3.1.4. MMI SET ..........................................................................................................16

4.3.2. Network ..................................................................................................................17

4.3.2.1. WAN Config .....................................................................................................17

4.3.2.2. LAN Config .......................................................................................................18

4.3.2.3. Qos Config .......................................................................................................19

4.3.2.4. Service Port .....................................................................................................21

4.3.2.5. DHCP SERVER ..................................................................................................21

4.3.2.6. SNTP .................................................................................................................23

4.3.3. VOIP ........................................................................................................................24

4.3.3.1. SIP Config .........................................................................................................24

SVP2000PP Business POE IP Phone User Manual

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4.3.3.2. IAX2 Config ......................................................................................................26

4.3.3.3. Stun Config ......................................................................................................27

4.3.3.4. DIAL PEER setting ............................................................................................28

4.3.4. Phone ......................................................................................................................32

4.3.4.1. DSP Config .......................................................................................................32

4.3.4.2. Call Service ......................................................................................................33

4.3.4.3. Digital Map Configuration ..............................................................................34

4.3.4.4. Phone Book .....................................................................................................35

4.3.4.5. Function Key ....................................................................................................36

4.3.5. Maintenance ..........................................................................................................38

4.3.5.1. Auto Provision .................................................................................................38

4.3.5.2. Syslog Config ...................................................................................................38

4.3.5.3. Config Setting ..................................................................................................39

4.3.5.4. Update .............................................................................................................40

4.3.5.5. Account Config ................................................................................................40

4.3.5.6. Reboot .............................................................................................................41

4.3.6. Security ...................................................................................................................42

4.3.6.1. MMI Filter ........................................................................................................42

4.3.6.2. Firewall ............................................................................................................42

4.3.6.3. NAT Config .......................................................................................................44

4.3.6.4. VPN Config .......................................................................................................46

4.3.7. Logout .....................................................................................................................48

5. C

ONFIGURATION VIA KEYPAD

......................................................................................... 48

5.1. Keypad introduction .................................................................................................48

5.2. Menu Tree .................................................................................................................48

6. APPENDIX .................................................................................................................. 49

6.1. S

PECIFICATION

..................................................................................................................49

6.1.1. Hardware ................................................................................................................49

6.1.2. Voice features ........................................................................................................49

6.1.3. Network features ...................................................................................................50

6.1.4. Maintenance and management ............................................................................50

6.1.5. Special features ......................................................................................................50

6.2. D

IGIT-CHARACTER MAP TABLE

.............................................................................................50

7.COPYRIGHT NOTICE..................................................................................................... 50

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone] /Fax:[phone] / Email: [e-mail] / Url: www.stephen-tele.com

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1. Introducing SVP2000P VoIP Phone

1.1. Thank you for your purchasing SVP2000P

Thank you for your purchasing SVP2000P, SVP2000P 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
The User Manual (you may download from our website)
IP Phone are designed to look like conventional phones, the following photo shows a broad overview of the IP Phone.

1.3. Keypad
Key Key name Function Description

Navigation

Navigation key assist users for operating...
In idle state they have special function
Left: Checking Incoming call
Up: Checking Missed Call
Right: Checking line status
Down: Checking IP info
OK: Enter into the phone's menu

Release key

Press RLS, the phone will skip to stand-by mode.

Note: DO NOT Press RLS during the configuration process, or else phone will not

save the configuration modified and return to stand-by status.

Phone Book

Access to phone book, check the record list and add new records and revise the
record. When check the phone book record, press this key again will return to idle
mode.

Envelope

LED inside, if blinks remind user have new voicemail.

Hold

Temporarily hold the active call during the talking; press the key again might
release the call. (Please refer to 3.5. - call hold for more details). In idle mode,
press this key, LCD will show “Do Not Disturb”, and this phone is set to be No
disturbing mode. Press this key again to cancel this function.

Transfer

Use the key to realize blind transfer or attended transfer

please refers to 3.4.-call

transfer for more details). In the idle mode, press this key, LCD will show”call

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forward”. After this indication

no. of SIP1 or SIP2. Press Soft2(ON) to enabled call forward function and set it to

always mode; Press Soft1(OFF) to close the function of forward

mute

Press this key in calling mode, you can hear the oth

not hear you

Line1/2/3

Here is Three SIP lines, user could select any one to make the call ,if it has been

registered

Volume -/+

Turn down or turn up the volume by pressing these two keys

Redial

1,In the hook off /hands

2,In stand

Hands-free

Make the phone into hands

Indicator light

If the light blinking, indicate the

Memory key

(1-6)

Users could store their commonly used number in these keys, and call for them as

speed dial.

Soft key 1/2/3

Keys combination, include functions such as SMS / SDial /Memo /Answer /Conf

/enter /save / quit /edit

1.4. Port for connecting

Port

Port name

Power switch

WAN

LAN

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disappears, User can configure the forward phone

-

free mode, use the key to dial the last call number;

-

by mode, it has a function to check the OUTGOING CALL

-free mode.

phone has missed call

/redial / and so on.

description

Input: 5V AC, 1A

10/100M Connect it to Network

10/100M Connect it to PC

POE IP Phone User Manual

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er side, and the other side can

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Handset

Headset

Headset

SVP2000P

provide two Ethernet ports and a power adaptor. Also has two headset interfaces with RJ

Please refer to safety notes of this manual carefully before power adaptor is connected.

2. Initial connecting and Setting

2.1. Connect the phone

2.1.1. Connect to network

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.

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

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 h

andset port by the handset cable in the package.

Step 3:

connect the power supply plug to the AC 5V adapter port on the back of the phone. Use the power cable to

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Port type: RJ-9 connector

Port type: RJ-9 connector

Port type: 3.5mm jack

back of your phone to the Ethernet port in your workspace.

POE IP Phone User Manual

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-9 port and 3.5mm jack.

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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 “Initializing

wait logon”. Later, a ready screen typically displays the date, time.

If your LCD screen displays different information from the above, yo

your network online mode.

If your VoIP phone registers into corporate IP telephony Server, your phone is ready to use.

2.1.2. Power adaptor connection

Make sure that the power you use is comply wit

1. Plug power adaptor to power socket.

2.

Plug power adaptor’s AC output to the AC5V port of

3.

There will be displayed black line and “initializing... wait logon...” on the screen. After finishing startup, p

greeting, current date and time and so forth.

4.

If phone has registered to the server, you can place or answer calls.

2.2. Basic Initialization

SVP2000P

is provided with a plenty of functions and parameters for configuration. User needs some network and VoIP

knowledge so that user could understand the meanings of parameters. In order to make user use the phone more easily

and convenient, there are basic c

onfigurations introduced which is mandatory to ensure phone calls.

2.2.1. Network settings

Make sure that network is connected already before setting network of phone.

configurations, so phone could access to network as long

phone has to be changed WAN network setting to Static IP or PPPoE.

Note:

during setting network parameter in menu, Please don’t press the

conf

iguration and will return to standby interface.

Setting PPPoE mode (for ADSL connection)

1. Get PPPoE account and password first.

2. Press , and press

twice, screen will show “

will display “WAN”.

3. Press Soft2 (Enter), then press

, choose

4. Press Soft1 (Edit), the screen will display “

(Del) to delete it, then input your PPPoE account and press Soft2 (Save). W

password settings,

5. Press Soft2 (Edit) again, then input your PPPoE password and press Soft2 (OK), With “Saved” displayed, screen will

display the current password: ******, and

confirm

save the Account and password.

6. Press Soft3 (Quit) once return to “

Net Mode

the current net mode.

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u need refer to the next section “Initial setting” to set

h the parameters of power adaptor.

SVP2000P to start up.

SVP2000P

uses DHCP to get WAN IP

as there is DHCP server in it. If there is no DHCP server available,

key, or else the phone will not save the

Network

”. Then press Soft2 (Enter) or "OK", the LCD screen

PPPoE.

Account

”. The screen will show the current account information. Press Soft1

ith “saved” displayed, screen will jump to

confirm

: you need input the passwo

rd again, after confirm, press soft2 (OK) to

”. Press Soft2 (Save) the screen will show “Saved” and then jump to show

POE IP Phone User Manual

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hone will show

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7. Press or

Soft3 (Quit) thrice return to standby, at this time the phone is trying to change to PPPoE mode. Press

for checking the status. If the screen shows “

PPPoE Server; if it shows an IP addres

s, then the phone has already get IP with PPPoE.

Setting Static IP mode (static ADSL/Cable, or no PPPoE / DHCP network)

1. Prepare the network’s parameters first, such as IP Address, Net mask, Default Gateway and DNS server IP address. If you

don’t know

this information, please contact the service provider or technician of network.

2. Press , then press

twice, chooses “

3. Press "OK" or Soft2 (Enter), then choose “

4. Press Soft1Edit) and screen will show “IP

”, then press Soft1 (Del) to delete. Input your IP address and press Soft2 (Save)

to save what you input. After “Saved

” shown, the screen will jump to show the

5. Press Soft1 (Del) to delete. Input

your Net mask and press Soft2 (Save). After “

show the Gateway information

6. Press Soft1 (Del) to

delete, Input your gateway and press

show the DNS information.

7. Press Soft1 (

Del) to delete. Input your DNS server address and press Soft2 (Save). After “

return to show IP information.

8. Press Soft3 (Quit) once,

the screen shows”

shows “Saved

” and then shows the current net mode.

9. Press

or Soft3 (Quit) thrice, return to main interface and at this time the phone is trying to change to Static

mode. Press

button, the screen shows

just now, if the phone could display

the right time, it shows that Static IP mode takes effect.

Setting DHCP mode

1. Press , then press

twice,

“WAN”.

2. Press "OK" or Soft2 (Enter) to show “

Net Mode

jump to show the current net mode.

3. Press or

Soft3 (Quit) thrice back to main interface and at this time phone is trying to change to DHCP mode.

Press until the phone shows “

DHCP

shows that DHCP mode takes effect.

3. SVP2000P’s basic operation

3.1. Answer calls

SVP2000P

will ring to indicate you when there is call incoming

 Answer with hook off

Take handset, you can talk directly. You can just hang up to finish talk

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Negotiating…

” it shows that the phone is trying to access to the

Network

”. Press "OK" or Soft2 (Enter), LCD screen will display “

Static”.

Net mask

information.

Saved

” shown, the screen will jump to

Soft2 (Save). After “Saved

” shown, the screen will jump to

Saved

Net Mode”. the cursor stay at“<>Static

”; with Soft2(Sa

“Static” .the screen shows

the IP address and gateway which were set

chooses “Network”. Then press "OK" or Soft2 (

Enter), the screen will

”. Select “DHCP”. Press Soft2 (Save), with

“

”,

If the screen shows the IP address and gateway which were set just now, it

,below is ways to answer call

：

。

POE IP Phone User Manual

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WAN”.

information.

” shown, the screen will

ve) pressed,the screen

show

saved” shown, screen will

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 Answer with hands-free

Press to begin talking. Press

again to finish talk

 Answer with headset

Press to answer the call, if

the phone detect headset LED will turn to

change to speaker mode. You may press

 Using hands-

free instead of handset during a talk

Press

and hook on the handset when you use handset to speak and want to change to use hand

Press again to finish talk.

 Using handset instead of hands-

free during a talk

Hook off the h

andset when you want to use hand

talk.

3.2. Place calls

 Using handset

Hook off (screen will show the current using line, or you could press key L1

could begin to dial number. After finishing it, press # and

hear a ring-back

tone and screen shows the callee’s number, it shows that the person you called is ringing. If callee answers

the call, you can begin to talk and your phone will keep showing callee’s number and counting time. Just hang up to finish

talk.
 Using hands-free

Press

（

screen will show the current

using line, or you could press key L1

could begin to dial number. After finishing it, press # and

hear a ring

back tone and screen shows the callee’s number, it shows that the person you called is ringing. If callee answers

the call, you can begin to talk and your phone will keep showing callee’s number and counting time. Press

finish talk.
 Using directory

Press in stand-

by mode, you will access to phonebook. If there are many persons records stored in the directory,

you can use &

to select number or press the first character of the name for searching the person which

you want to contact. Press

to forward

number shown on the screen.
 Speed dial

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green, when

you press

or Soft3 (Close) to finish the call.

s-

free to speak and want to change to use handset. Just hook on to finish

-

L3 to select), after getting dialing tone, you

SVP2000P

will send the number and call the number. When you

using line, or you could press key L1

-

L3 to select), after getting dialing tone, you

SVP2000P

will send the number and call the number. When you

,and press

to backward. Press Soft2 (Dial) to dial the current

POE IP Phone User Manual

- 10 -

again the phone

s-free to speak.

again to

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Speed dial means user can make calls directly without hook off or using hands-free. User can dial number in stand-by mode,
but first, user need to add and edit SDial no. By pressing Soft2 (SDial) to edit and save the number to be a SDial number. In
this way, user could make a call only press the number and Soft3 (Dial).
 Multi-line calls
SVP2000P supports 3 SIP lines, that is user could use 3 SIP accounts to register and make calls. System will use SIP 1 as
default line to call.
There are most two calls at the same time. Screen will display the incoming call number when user is keep talking. You can
press Soft1 (Answer) to accept it, and hold the first one (if you want to use this function, you need enable Call Waiting of
the phone first). Use Soft1 (Switch) to switch the two calls to talk.

3.3. End calls

 Hang up with handset hook on
Hook on to finish talking.
 Hang up with hands-free

Press to finish talk when phone is in hands-free status.

Note: user can not finish talk by pressing if phone is used handset to talk.

 Hang up a active call with 2 calls
When there are two calls,user might use Soft1(Switch)to switch to the call you want to hang up first. Then press Soft3
(Close) to finish talk, and phone will switch to the other call automatically.
Note：it is no use to press Soft3(Close) to finish talk, if there is only one current call.

3.4. Call transfer



Blind Transfer

During talk, press or Soft2 (Transf), and then dial the number that you want to transfer to, and finished by "#".

Phone will transfer the current call to the third party. After finishing transfer, the call you talk to will be hanged up. User
can not select SIP line when phone transfers call.



Attended Transfer

During talk, press or Soft2 (Transf), then input the number that you want to transfer to and press Soft2 (Send).

After that third party answers, then press to complete the transfer. (You need enable call waiting and call transfer

first). If there are two calls, you can just talk to one, and keep hold to the other one. The one who is keep hold can not
speak to you or hear from you. In other way, if user wants to invite the third party during the call, they can press Soft1
(Conf) to make calls mode in conference mode. If user wants to stop conference, user can press Soft1 (Split). (User must
enable call waiting and three way call first).
Note: the server that user uses must support RFC3515 or it might not be used



Alert Transfer

During the talk, press or Soft2 (Transf) firstly, then press Soft2 (Send) after inputting the number that you want to

transfer. You are waiting for connection, now, press or Soft2 (Transf) and the transfer will be done. (To use this

feature, you need enable call waiting and call transfer first)

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3.5. Call hold

During talking, user could press

to hold the current call. Press

active.

3.6. 3-way conference call

User can press Soft1 (Conf) to dial the line2 (press

talking with line1. After line2 connect, user can press Soft2 (Conf) to enter into conference mode. To back to line1 from

conference, please press Soft1 (Split); to end the call, please press Soft3 (Close) or

3.7. Switchboard Operator feature

User can press Soft1 (Conf) to dial the

line2 (

talking with line1. After line2 connect, user can press Soft1 (Switch) to select which line you prefer to

to input the number you want to transfer and press

3.8. Call records

SVP2000P

supports 100 items of missed call, 100 items of incoming call, and 100 items of dialed call. If the records are full,

the newes

t will replace the oldest. If phone’s power cut or reboot, call records will be discarded.

 Missed call

Press

, and screen displays “Missed Call” with the number and time of missed call. User can also use

to browse the missed call records, or press Sof

(EDial) again to change the current number. Pressing

number. If there is no missed call, screen will show “List Is Empty”

 Incoming call

Press

and screen displays “Incoming Call

(Option) to check the details of this record, then press Soft2 (EDial) again to change the current number. Pressing Soft2

(Dial) will call this number directly if user don’t modify the number. If there is no incoming call

Empty”.

 Dialed call

Press , and use &

to browse the dialed call

record, then press Soft2 (EDial) again to change the current number. Pressing Soft2 (Dial) will call this number direc

user don’t modify the number. If there is no dialed call, screen will show “List Is Empty”.

4. Web configuration

4.1. Introduction of configuration

4.1.1. Ways to configure

SVP2000PP

Business

[e-mail] / Url: www.stephen-

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again to

return

Soft1 (

Answer) to answer the call directly if this cal

press

press Soft1 (

Answer) to answer the call directly if this call is from line2) during

again to do the transfer.

t1 (Option) to check the details of this record, then press Soft2

Soft2 (

Dial) will call this number directly if user don’t modify the

.

”, by pressing &

to browse the

records; or

press Soft1 (Option) to check the details of this

POE IP Phone User Manual

- 12 -

the call or switch the call

l is from line2) during

.

transfer, then press

&

records; or press Soft1

,screen will show “List Is

tly if

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SVP2000P

has three different ways to different users.

 Use phone keypad.
 Use web browser

（

recommendatory way

 Use telnet with CLI command.

4.1.2. Password Configuration

There are two levels to access to phone

configuration parameters, while user with general level can set all configuration parameters except SIP (1

some parameters can not be changed, such as server addre

username and password.

 Default user with general level：

 username：guest
 password：guest

 Default user with root level：

 username：admin
 password：admin

The default password of phone screen menu is

4.2. Setting via web browser

When this phone and PC are connected to 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

If you do not know the IP address, you can l

The login page is as below picture

※

:

After you configure the ip phone, you need click save button in config under Maintenance in the left catalog to save

your configuration. Otherwise the phone wil

4.3. Configuration via WEB

4.3.1. BASIC

4.3.1.1. Status

SVP2000PP

Business

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).

:

root level and general level. User with root level can browse and set all

ss and port. User will has different access level with different

123.

:xxxx/).

ook it up on the phone’s display by pressing

l lose your modification after power off and on.

POE IP Phone User Manual

- 13 -

-2) or IAX2’s that

button.

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Field name

Network

Shows the configuration information on WAN and LAN port, including the connect

mode of WAN port (Static,

LAN port, ON or OFF of DHCP mode of LAN port.

Phone Number

Shows the phone numbers provided by the SIP LINE 1

The last line shows the version number and issued date.

4.3.1.2. Wizard

Field Name

Please select the proper network mode according to the network condition.

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 fro

information artificially.

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Status

Explanation

DHCP, PPPoE), MAC address, the IP address of WAN port and

-

3 servers and IAX2.

Wizard

Explanation

SVP2000P

provide three different network

m the DHCP server automatically; need not to input this

POE IP Phone User Manual

- 14 -

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 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)simply, also can browse too. Click

【【【【

BACK

】】】】

can return to the last page.

Static IP Address Input the IP address distributed to you.
Netmask Input the Netmask distributed to you.
Gateway Input the Gateway address distributed to you.
DNS Domain Set DNS domain postfix. When the domain which you input can not be parsed, phone will

automatically add this domain to the end of the domain which you input before and parse

it again.
Primary DNS Input your primary DNS server address.
Alter DNS Input your standby DNS server address.

Display Name Set the display name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
User Name Input your SIP register account name.
Password Input your SIP register password.
Phone Number Input the phone number assigned by your VOIP service provider.
Enable Register Start to register or not by selecting it or not.

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Display detailed

information that you manual config.

Choose DHCP MODE,click

【【【【

NEXT

】】】】

can config SIP(default SIP1)simply, also can browse too. Click

last page. Like Static IP MODE。
Choose PPPoE MODE,click

【【【【

NEXT

】】】】

can config the PPPoE account/password and SIP(default SIP1)simply, also can browse

too. Click

【【【【

BACK

】】】】

can

return to the last page. Like Static IP MODE

PPPoE Server

It will be provided by ISP.

Username

Input your ADSL account.

Password

Input your ADSL password.

Notice: Click

【【【【

Finish

】】】】

button after finished your setting, IP Phone will save the setting

reboot, you can dial by the SIP account.

4.3.1.3. Call Log

You can query all the outgoing through this page.

Field name

Start Time

Display the start time of the outgoing record.

Last Time Display the

conversation time of the outgoing record.

Called Number

Display the account/protocol/line of the outgoing record.

4.3.1.4. MMI SET

Field name

Language Set

Set the language of phone, English is default.

SVP2000PP

Business

[e-mail] / Url: www.stephen-

tele.com

。

automatically and reboot, After

Call Log

explanation

MMI SET

explanation

POE IP Phone User Manual

- 16 -

【【【【

BACK

】】】】

can return to the

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Greeting Message

The greeting message will display on lcd when phone is idle. It can support 16 chars. the
default chars are VOIP PHONE.

4.3.2. Network

4.3.2.1. WAN Config

WAN Config

Field Name explanation

Active IP The current IP address of the phone.
Current Netmask The current Netmask address.
MAC Address The current MAC address of the phone.
Current Gateway 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. SVP2000P 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.

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- 18 -

Obtain DNS server

automatically

Select it to use DHCP mode to get DNS address, if you don’t select it, you will use static DNS
server. The default is selecting it.

If you use static mode, you need set it.
IP Address Input the IP address distributed to you.
Netmask Input the Netmask distributed to you.
Gateway Input the Gateway address distributed to you.

DNS Domain

Set DNS domain postfix. When the domain which you input can not be parsed, phone will
automatically add this domain to the end of the domain which you input before and parse it

again.
Primary DNS Input your primary DNS server address.
Alter DNS Input your standby DNS server address.

If you uses PPPoE mode, you need to make the above setting.
PPPoE Server It will be provided by ISP.
Username Input your ADSL account.
Password Input your ADSL password.
Notice:

1)Click “Apply” button after finished your setting, IP Phone will save the setting automatically and new setting will take

effect.

2)If you modify the 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 DHCP server distributed is same as network ID which is used by LAN of system, system 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 system uses DHCP client to get IP in startup; if system uses DHCP client to get IP in running status
and network ID is also same as LAN’s, system will refuse to accept the IP to configure WAN. So WAN’s active IP will be

0.0.0.0

4.3.2.2. LAN Config

LAN Config

Field name explanation

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LAN IP Specify LAN static IP.
Netmask Specify LAN Netmask.

DHCP Service

Select the DHCP server of LAN port or not. After you 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.

NAT Select NAT or not.

Bridge Mode

Select Bridge Mode or not: If you select Bridge Mode, the phone will no 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.

In chart 1, there is a layer 2 switch without setting VLAN. Any broadcast frame will be transmitted to the other ports except

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the send port. For example, a broadcast information is sent out from port 1 then transmitted to port 2,3and 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 explanation

VLAN Enable Before select it to enable VLAN, you need enable Bridge mode in LAN config.

VLAN ID Check Enable

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.

Voice/Data VLAN
differentiated

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 untagged means after using VLAN, only

VoIP packets will add voice VLAN ID. Other data packets will not use VLAN.
DiffServ Enable Select it or not to Enable or disable DiffServ.
DiffServ Value Set DiffServ value, the common value is 0x00.
Voice 802.1P Priority Specify 802.1P Priority of voice/signal data package.
Data 802.1P Priority 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 disables 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 enables 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 untagged, then the packet of the signal/voice will use the

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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) One must to 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 not match 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.

SERVICE PORT

Field name explanation

HTTP Port

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
Telnet Port 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
RTP Initial Port 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

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DHCP SERVER

Field name explanation

DHCP Leased Table 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.

Shows the DHCP Lease Table, the unit of Lease time is Minute.
Lease Table Name Specify the name of the lease table

Start IP Set the start IP address of the lease table

End IP

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.

Netmask Set the Netmask of the lease table
Gateway Set the Gateway of the lease table
Lease Time Set the Lease Time of the lease table
DNS Set the default DNS server IP of the lease table; Click the Add button to submit and add this

lease table

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 enabled. 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.

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- 23 -

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 explanation

Server Set SNTP Server IP address.
Time Zone Select the Time zone according to your location.
Time Out Set the time out, the default is 60 seconds.
12 Hours Systems Switch the time mechanism between 12 hours and 24 hours.

Default is 24 hours mode
SNTP 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 Setup start and end week
Day Setup start and end day
Hour Setup start and end hours
Minute Setup start and end minutes

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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.

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Business

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tele.com

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SIP Config

Field name explanation

Choose line to set info about SIP, there are 3 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;
Server Name Set the server name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
Account Name Input your SIP register account name.
Password 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

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.)
Proxy Server Port Set your Proxy SIP server port.
Proxy Username Input your Proxy SIP server account.
Proxy Password Input your Proxy SIP server password.

Domain Realm

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).
Enable Register Start to register or not by selecting it or not.

Register Expire Time

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.
NAT Keep Alive Interval Set examining interval of the server, default is 60 seconds
User Agent Set the user agent if have, the default is VoIP Phone 1.0
Signal Key Set the key for signal encryption
Media Key Set the key for RTP encryption
Local port Set sip port of each line
Ring type Set ring type of each line
Hot line Number Set hot line number of each line
Conference Number Configure conference number in server conference.
Transfer Expire Time For the phone supports the transfer of certain special features server, set interval time

between sending “bye” and hanging up after the phone transfers a call.
Enable subscribe Enable the option ,the phone will receive the notify from the server.
Enable Keep
Authentication

Enable/Disable Keep Authentication System will take the last authentication field which is
passed the authentication by server to the request packet. It will decrease the server’s
repeat authorization work, if it is enable.

NAT Keep Alive

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.

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Enable Via rport Enable/Disable system to support RFC3581. Via rport is special way to realize SIP NAT.
Enable PRACK Enable or disable SIP PRACK function, suggest use the default config.
Long Contact Set more parameters in contact field; connection with SEM server
Enable URI Convert 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;
Enable DNS SRV Support DNS looking up with _sip.udp mode

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

Select DTMF sending mode, there are three modes:

 DTMF_RELAY

 DTMF_RFC2833

 DTMF_SIP_INFO

Different VoIP Service providers may provide different modes.

RFC Protocol Edition

Select SIP protocol version to adapt for the SIP server which uses the same version as you

select. For example, if the server is CISCO5300, you need to change to RFC2543, else phone

may not cancel call normally. System uses RFC3261 as default.
Transport Protocol Set transport protocols, TCP or UDP;
RFC Privacy Edition Set Anonymous call out safely; Support RFC3323and RFC3325;
Subscribe Expire Time Overtime of resending subscribe packet. Suggest using the default config.
Enable Conference
number

Set to use sever conference.

MWI Number Input the number of the server's voice-mail box
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 receives the packets sent from server, phone will

use the source IP address, not the address in via field.
Enable GRUU Set to support GRUU
Enable Display name
Quote

Set to make quotation mark to display name as the phone sends out signal, in order to be

compatible with server.

4.3.3.2. IAX2 Config

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IAX2 Config

Field name explanation

Register Status Shows if the phone has been registered the IAX2 server or not.
IAX2 Server Addr Input your IAX2 server address.
IAX2 Server Port Set your IAX2 server port, the default is 4569.
Account Name Input your IAX2 register account name.
Account Password Input your IAX2 register password.
Phone Number Input your assigned phone number (usually it is same you’re your IAX2 account name).
Local Port Set your local sport,the default is 4569.
Voice Mail Number 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.
Enable G.729 Enable or disable code G.729 by selecting it or not

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.

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STUN

Field name explanation

STUN NAT Transverse Shows STUN NAT Transverse estimation, true means STUN can penetrate NAT, while False

means not.
STUN Server Addr Set your SIP STUN Server IP address
STUN Server Port Set your SIP STUN Server Port

STUN Effect Time

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.
Local SIP Port Set the SIP port.

Choose line to set info about SIP, There are 3 lines to choose. You can switch by

【【【【

Load

】】】】

button.

Use Stun Enable/Disable SIP STUN.
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

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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 functions:

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.

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DIAL PEER

Field name explanation

Phone number

There are two types of matching conditions: one is full matching, the other is prefix matching.
In the Full matching, you need input your desired phone number in this blank, and then you
need dial the phone number to realize calling to what the phone number is mapped. In the
prefix matching, you need input your desired prefix number and T; then dial the prefix and a
phone number to realize calling to what your prefix number is 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 on SIP2 line, you
need input 255.255.255.255 or 0.0.0.2 in it.SIP3 into 0.0.0.3

Port Set the Signal port, the default is 5060 for SIP.
Alias Set alias. This is optional config item. If you don’t set Alias, it will show no alias.
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 different signal protocol, SIP or IAX2
Suffix Set suffix, this is optional config item. It will show no suffix if you don’t set it.
Delete Length Set delete length. This is optional config item. For example: if the delete length is 3, the phone

will delete the first 3 digits then send out the rest digits. You can refer to examples of different
alias application to know how to set delete length.

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Examples of different alias application

Set by web explanation example

You need set phone number,
Destination, Alias and Delete
Length.
Phone number is XXXT;
Destination is

255.255.255.255 (0.0.0.2) and
Alias is del.
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.

If you dial “93333”, the SIP2 server will receive
“3333”

This setting will realize speed
dial function, after you dialing
the numeric key “2”, the
number after all will be sent
out.

When you dial “2”, the SIP1 server will receive
33334444

The phone will automatically
send out alias number adding
your dialed number, if your
dialed number starts with your
set phone number.

When you dial “8309“, the SIP1 server will receive
“07558309”

You need set Phone Number,
Alias and Delete Length.
Phone number is XXXT and
Alias is rep:xxx
If your dialed phone number
starts with your set phone
number, the first digits same
as your set phone number will
be replaced by the alias
number specified and New
phone number will be send

When you dial “0106228”, the SIP1 server will
receive “86106228”

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out.

If your dialed phone number
starts with your set phone
number. The phone will send
out your dialed phone number
adding suffix number.

When you dial “147”, the SIP1 server will receive
“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
Second Codec The second preferential DSP codec: G.711A/u, G.722, G.723, G.729
Third Codec The third preferential DSP codec: G.711A/u, G.722, G.723, G.729
Forth Codec The forth preferential DSP codec: G.711A/u, G.722, G.723, G.729
Fifth Codec The fifth preferential DSP codec: G.711A/u, G.722, G.723, G.729
Input Volume

Specify Input (MIC) Volume grade.

；

Hands-free Volume Specify Hands-free Volume grade
G729 Payload Length Set G729 Payload Length

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Handdown Time

Specify the least reflection time of Handdown, the default is 200ms.

Ring Type

Select Ring Type

Output Volume Specify Output

(receiver) Volume grade.

Ring Volume

Specify Ring Volume grade

G722 Timestamps 160/20ms or

320/20ms is available

G723 Bit Rate

5.3kb/s or 6.3kb/s is available

Default Ring Type

Set up the ring by default

Signal Standard

Select Signal Standard.

VAD

Select it or not to enable or disable VAD.

over 20ms.

4.3.4.2. Call Service

In this web page, you can configure

Hotline, Call Transfer,

Field name

Hotline

Specify Hotline number. If you set

No Answer
Time

Specify No Answer Time

P2P IP Prefix

Set Prefix in peer to peer IP call. For example

IP Prefix as 192.168.1., you dial only #119 to

means to disable dialing IP.

Remote Record
No

Set Remote Record

number

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.

Ban Outgoing

If you select Ban Outgoing to enable it, and you can not dial out any number.

Enable Call
Transfer

Enable Call

Transfer by selecting it.

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If enable VAD, G729 Payload length could not be

Call Waiting

, 3 Ways Call, Black List, white list Limit List and so on.

Call Service

explanation

the number, you can not dial any other numbers.

:

what you want to dial is 192.168.1.119, If you define P2P

reach 192.168.1.119. Default is “.”. If there is no “.” Set, it

. Via dialing this number, you can hear

all voice records in your VoIP server.

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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 If select it, the phone will auto answer when there is an incoming call.
Use Record
Server

Select it or not to Enable or disable Use Record Server.

Auto
handdown

The phone will hang up and return to standby automatically at hands-free mode

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

Limit List

Set Add/Delete Limit List. Please input the prefix of those phone numbers which you forbid the phone to
dial out. For example, if you want to forbid those phones of 001 as prefix to be dialed out, you need
input 001 in the blank of limit list, and then you 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 system 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.

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 totally to the 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.

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Digital Map Configuration

Field name explanation

End with "#" Set Enable/Disable the phone ended with “#” dial.
Fixed Length Specify the Fixed Length of phone ending with.

Time out

Set the timeout of the last dial digit. The call will be sent after timeout.

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.

Cause extensions 1000-8999 to be dialed immediately
Cause 8 digit numbers started with 9 to be dialed immediately
Cause 911 to be dialed immediately after it is entered.
Cause 99 to be dialed after 4 seconds.
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.

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Phone Book

Field name explanation

Shows the detail of current phonebook.
Name Shows the name corresponding to the phone number
Number Shows the phone number
Ring Type 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

4.3.4.5. Function Key

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Field name

Contrast

Set contrast of screen

Luminance

Set luminance of screen

MWI Number

To listening record in server , we defined the

or hands-

free, then press

Line: select SIP1, SIP2, SIP3, Dial peer

, or IAX2 in function key type. After you set it, you pick up handset or hands

press this

function key, then you can use the corresponding IP line.

Memory Key:

you can set a number for each memory key. After set it, you can dial the number you set by pressing this

memory key.
Key event: function mode
Remark:
 You can set speed dial function

by Memory Key mode.

For example, you need set speed dial 8000 via sip 1.

Select memory

key in F4’s function key type, then fill 8000@1/f in the corresponding right table.

 You can set shortcut

key of pbook, redial, DND, MWI, call forward, or callers by

Select key event in function key type, then fill F_PBOOK, F_REDIAL, F_DND, F_MWI, F_CFWD,or F_CALLERS in the

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Function Key

explanation

function key F10 ,After you set it, you can pick up

to listen record in server.

K

ey Event mode in function key type.

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corresponding right table.
For example:

4.3.5. Maintenance

4.3.5.1. Auto Provision

Auto Provision

Field name explanation

Current Config Version 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 Set FTP server Password.
Config File Name 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 Input the Encrypt Key, if the configuration file is encrypted.
Protocol Type

Select the Protocol type FTP、TFTP or HTTP.
Update Interval Time Set update interval time, unit is hour.

Update Mode

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.

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Level 2---critical: Your system has serious problem.
Level 3---error: The error will affect your system working.
Level 4---warning: There are some potential dangers. But your system can work.
Level 5---notice: Your system works well in special condition, but you need to check its working environment and parameter.
Level 6---info: the daily debugging info.
Level 7---debug: the lowest debug info. Professional debugging info from R&D person.
At present, the lowest level of debug information send to Syslog is info, debug level only can be displayed on telnet.

Syslog Configuration

Field name explanation

Server IP Set Syslog server IP address.
Server Port Set Syslog server port.
MGR Log Level Set the level of MGR log.
SIP Log Level Set the level of SIP log.
IAX2 Log Level Set the level of IAX2 log.
Enable Syslog Select it or not to enable or disable syslog.

4.3.5.3. Config Setting

Config Setting

Field name explanation

you can save all changes of configurations. Click the Save button, all changes of configuration

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Save Config will be saved, and be effective immediately. .
Backup Config Right clicks on “Right click here…” and select “Save Target As….” then you will save the config

file in .txt format

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 explanation

Web Update

Click the browse button, find out the config file saved before or provided by manufacturer,

download it to the phone directly, press “Update” to save. You can also update downloaded

update file, logo picture, ring, mmiset file by web.
Server Set the FTP/TFTP server address for download/upload. The address can be IP address or

Domain name with subdirectory.
Username Set the FTP server Username for download/upload.
Password Set the FTP server password for download/upload.
File name 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

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Account Configuration

Field name explanation

Keyboard Password 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

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..

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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 Filter 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.

MMI Filter 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.

4.3.6.2. Firewall

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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 Select it to Enable out_ access rule
Input/Output 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
Src Addr Set source address. It can be single IP address, network address, complete address 0.0.0.0, or

network address similar to *.*.*.0

Des Addr Set the destination address. It can be IP address, network address, complete address 0.0.0.0,

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or network address similar to *.*.*.*

Src Mask

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.

Des Mask

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.

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.

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

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NAT Configuration

Field name explanation

IPSec ALG It is an encryption technology. Select it to enable IPSec ALG, the default is enable

FTP ALG

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

PPTP ALG Select it enable PPTP ALG, the default is enable

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Shows the NAT TCP mapping table

Shows the NAT UDP mapping table

Transfer Type Select the NAT mapping protocol style, TCP or UDP
Inside IP 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.

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.

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone] /Fax:[phone] / Email: [e-mail] / Url: www.stephen-tele.com

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VPN Configuration

Field name explanation

VPN IP 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；

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone]

/Fax:[phone] / Email:

VPN Server Addr

Set VPN Server IP Address

VPN Server Port

Set VPN Server Port

VPN Server Addr

Set VPN L2TP Server IP address

VPN User Name

Set User Name access to VPN L2TP Server

VPN Password Set

Password access to VPN L2TP Server

4.3.7. Logout

Click Logout,

and you will exit web page. If you want to enter it next time, you need input user name and password again.

5. Configuration via Keypad

5.1. Keypad introduction

User can browse, modify or cancel via screen menu by using

 Use need input password (default 123

),when login the menu (

5.2. Menu Tree

Menu Tree List

：：：：

SVP2000PP

Business

[e-mail] / Url: www.stephen-

tele.com

and

system config)

POE IP Phone User Manual

- 48 -

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone] /Fax:[phone] / Email: [e-mail] / Url: www.stephen-tele.com

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6. Appendix

6.1. Specification

6.1.1. Hardware

Model NO. SVP2000P
Power ?(Input/ Output) Input:100-240V / Output:5V/1A

Port WAN 10/100Base-T? RJ-45 for WAN

LAN 10/100Base-T? RJ-45 for PC

Power Consumption Idle: 2.5W/Active: 2.8W
Power Over Ethernet IEEE802.3af Power Over Ethernet on WAN

LCD Size 53.5 x 70mm
Operation Temperature 0-40℃

Relative Humidity 10-65%
Main Chipset Broadcom

SDRAM 128Mbits
Flash 32Mbits

6.1.2. Voice features

 SIP supports 3 SIP servers
 Support SIP 2.0 (RFC3261) and correlative RFCs
 Codec: G.711A/u, G.723.1 high/low, G.729a/b,G.722,G.722.1,G.726
 Echo cancellation: G.168 Compliance in LEC, additional acoustic echo cancellation(AEC) can reach 96ms max filter

length in hands-free mode

 Support Voice Gain Setting, VAD, CNG
 Support full duplex hands-free
 SIP support SIP domain, SIP authentication(none basic, MD5), DNS name of server, Peer to Peer/ IP call
 Automatically select calling line, if one line can’t be connected, the phone can automatically switch to other line to call.
 DTMF Relay: support SIP info,DTMF Relay,RFC2833
 SIP application: SIP Call forward/transfer（blind/attended)/hold/waiting/3 way talking/
 Exchange /sms /pickup /joincall /redial /unredial /vport
 Call control features: Flexible dial map, hotline, empty calling No. reject service, black list for reject authenticated call,

limit call, no disturb, caller ID, Flexible deer peer rule.

 Support phonebook 500 records,Incoming calls / outgoing calls / missing calls. Each supports 100 records
 Support conference on server.
 Support IAX2
 Phonebook supports vcard standard
 12/24 hours time display
 Support daylight saving time
 Support path, gruu
 Support SIP Privacy

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone] /Fax:[phone] / Email: [e-mail] / Url: www.stephen-tele.com

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 Support SMS
 Support Memo
 Support WMI
 Support Speed dial
 Support Alarm clock

6.1.3. Network features

 WAN/LAN: support bridge and router model
 Support PPPoE for xDSL
 Support basic NAT and NAPT
 Support VLAN (optional: voice vlan/ data vlan)
 NAT Penetrate, Stun Penetrate
 Support DMZ
 Support VPN (L2TP) function
 Wan Port supports main DNS and secondary DNS server, can select dynamically to get DNS in DHCP mode or statically

set DNS address.

 Support DHCP client on WAN
 Support DHCP server on LAN
 QoS with DiffServ
 Network tools in telnet server: including ping, trace route, telnet client

6.1.4. Maintenance and management

 Upgrade firmware through POST mode
 Web ,telnet and keypad management
 Management with different account right
 LCD and WEB configuration can be modified into requested language, and support multi-language dynamically shifted
 Upgrade firmware through HTTP, FTP or TFTP Telnet remote management/ upload/download setting file
 Support Syslog
 Support Auto Provisioning (upgrade firmware or configuration file)


6.1.5. Special features

 Support 3 softkeys, 6 memory keys, Navigation key.
 RLS,Pbook,MWI,HOLD,Trans,Mute,L1-L3,Vol -/+,Redial

6.2. Digit-character map table

Keypad Character Keypad Character

1 @

7 P Q R S p q r s

2 A B C a b c

8 T U V t u v

3 D E F d e f

9 W X Y Z w x y z

4 G H I g h i

*/.

5 J K L j k l

0

SVP2000PP Business POE IP Phone User Manual

STEPHEN TECHNOLOGIES CO.,LIMITED

/ 5/F, Building NO.1, TongXin Industry Zone, HengGang, LongGang, Shenzhen, G.D, China, 518115

Tel: [phone] /Fax:[phone] / Email: [e-mail] / Url: www.stephen-tele.com

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6 M N O m n o

#/=

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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
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Co.,Limited assumes no responsibility for any errors in this document. Furthermore, Stephen Technologies
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within this document are subject to change at any time, without notice and without obligation to notify any
person of such change.