Chima SVG300S User manual

SVG300S 2 FXS SIP ATA User Manual
陈泽科技有限公司 www.stephen-tele.com 1
SVG300S 2 FXS SIP ATA User Manual
P
ST PH N T CHNOLOGI S CO.,LIMIT D
ALL RIGHTS R S RV D

SVG300S 2 FXS SIP ATA User Manual
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1 Introduction ..................................................................................................................................................... 3
1.1 Hardware Overview ............................................................................................................................. 3
1.2 Software Overview ............................................................................................................................... 3
2 IVR Interface for The Sip ATA ............................................................................................................................ 4
3 Setup the Sip ATA by Web Browser .................................................................................................................. 5
3.1 Login ..................................................................................................................................................... 5
3.2 System Information .............................................................................................................................. 5
3.3 Phone Book .......................................................................................................................................... 6
3.4 Phone Settings ..................................................................................................................................... 6
3.5 Network ............................................................................................................................................. 11
3.6 SIP Settings ......................................................................................................................................... 14
3.7 NAT Trans. .......................................................................................................................................... 18
3.8 Others ................................................................................................................................................ 18
3.9 System Auth. ...................................................................................................................................... 19
3.10 Save Change ....................................................................................................................................... 20
3.11 Update ............................................................................................................................................... 20
3.12 Reboot................................................................................................................................................ 21
4 Setup the Sip ATA by using Console (Hyper Terminal) ................................................................................... 21
4.1 Configuration ..................................................................................................................................... 21
4.2 Using CLI command to configure the Sip ATA .................................................................................... 22
5 How to make a phone call .............................................................................................................................. 28
6 Get a FWD account ........................................................................................................................................ 29

SVG300S 2 FXS SIP ATA User Manual
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Introduction
This user’s manual is for SVG300S Analog Telephone adapter (ATA). This user’s manual will explain the IVR instruction,
web configuration and command line configuration for the Sip ATA. Before using the Sip ATA, some setup processes are
required to make the Sip ATA work properly.
1.1 Hardware Overview
The Sip ATA has Networking interfaces, telephone interfaces, L D indication, power connector and Reset button.
1.1.1 Two RJ-45 Networking interfaces, these two interfaces support 10/100Mps Fast thernet. you can connect one
RJ-45 Fast thernet port to the ADSL or Switch, and connect the other one to your computer.
1.1.2 Two RJ-11 Type analog telephone jack interfaces. Both Phone1 and Phone2 are for user to connect to a phone set.
1.1.3 L D Indication: There are five L D indicators in the Sip ATA to show the Power, network, and Off-Hook indication.
1.1.4 Reset Button: Press Reset button for longer than 5 seconds to make the Sip ATA restore factory default settings and
reboot. If TA works improperly, please make an attempt to Reset and then configure again.
1.2 Software Overview
Network Protocol Tone
•
SIP v1 (RFC2543), v2(RFC3261)
•IP/TCP/UDP/RTP/RTCP
•IP/ICMP/ARP/RARP/SNTP
•TFTP Client/DHCP Client/ PPPo Client
•Telnet/HTTP Server
•DNS Client
•NAT/DHCP Server
•
Ring Tone
•Ring Back Tone
•Dial Tone
•Busy Tone
•Programming Tone
Phone Function
Codec
•
Volume Adjustment
•Speed dial key
•Phone book
•Flash
•
G.711: 64k bit/s (PCM)
•G.723.1: 6.3k / 5.3k bit/s
•G.726: 16k / 24k / 32k / 40k bit/s (ADPCM)
•G.729A: 8k bit/s (CS-AC LP)
•G.729B: adds VAD & CNG to G.729
IP Assignment
•
Static IP
•DHCP
•PPPo
Voice Quality
•
VAD: Voice activity detection
•CNG: Comfortable noise generator
•L C: Line echo canceller
•Packet Loss Compensation
•Adaptive Jitter Buffer
Security
•
HTTP 1.1 basic/digest authentication for Web
setup
•MD5 for SIP authentication (RFC2069/ RFC
2617)
QoS
C
all Function
•
ToS field
•
Call Hold
•Call Waiting
•Call Forward
•Caller ID
•3-way conference
NAT Traversal
•
STUN
Configuration
TMF Function
•
Web Browser
•Console/Telnet
•IVR/Keypad
•
In-Band DTMF
•Out-of Band DTMF
•SIP Info
Firmware Upgrade

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SIP Server
•
Registrar Server (three SIP account)
•Outbound Proxy
•
TFTP
•Console
•HTTP

IVR Interface for The Sip ATA
You can use the PSTN phone to configure the Sip ATA. Please follow the instruction to configure your terminal adapter.
Group
IVR Action
IVR Menu Choice
Parameter(s)
Note
s:
Function Reboot #195# None
After you hear “Option
Successful,” hang-up. The
system will reboot
automatically.
Function Factory Reset
#198# None
System will automatically
Reboot. WARNING: ALL
“User-Changeable”
NOND FAULT S TTINGS WILL
B LOST! This will include
network and service provider
data.
Info
Check IP
Address #120# None
IVR will announce the current
IP address of the Sip ATA
Info Check IP Type #121# None
IVR will announce if DHCP in
enabled or disabled.
Info
Check the
Phone Number
#122# None
IVR will announce current in
use VoIP number
Info
Check Network
Mask #123# None
IVR will announce the current
network mask of the Sip ATA.
Info Check Gateway
IP Address #124# None
IVR will announce the current
gateway IP address of the Sip
ATA.
Info
Check P
rimary
DNS Server
Setting
#125# None
IVR will announce the current
setting in the Primary DNS
field.
Info
Check
Firmware
Version
#128# None
IVR will announce the version
of the firmware running on the
Sip ATA.
Setting
Set DHCP
client #111# None
The sys
tem will change to
DHCP Client type
Setting Set Static IP
Address
#112xxx*xxx*xxx*xx
x#
nter IP address using numbers
on the telephone key pad. Use
the * (star) key when entering a
decimal point.
DHCP will be disabled and
system will change to the Static
IP type.
Setting Set Network
Mask
#113xxx*xxx*xxx*xx
x#
nter value using numbers on the
telephone key pad. Use the *
(star) key when entering a
decimal point.
Must set Static IP first.
Setting Set Gateway IP
Address
#114xxx*xxx*xxx*xx
x#
nter IP ad
dress using numbers
on the telephone key pad. Use
the * (star) key when entering a
decimal point.
Must set Static IP first.
Setting
Set Primary
#115xxx*xxx*xxx*xx
nter IP address using numbers
Must set Static IP first.

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DNS Server
x#
on the telephone key pad. Use
the * (star) key when entering a
decimal point.
Setting Set Codec #130+[1-8]#
1:G.711 u-Law, 2: G.711 a-Law, 3:
G.723.1, 4: G.729a, 5: G.726 16K,
6: G.726 24K, 7: G.726 32K, 8:
G.726 40K,
You can set the codec you want
to the first priority.
Setting Set Handset
Gain #131+[00~15]# Handset Gain from 0~15 You can set the Handset gain to proper value, default is 6
Setting Set Handset
Volume #132+[00~12]# Handset Volume from 0~12
You can set the Handset
volume to proper value,
default is 10
By default, NAT is on, LAN’s ip is 192.168.123.1(dial #120# to check LAN’s ip), WAN is DHCP client(dial #126# to check
WAN’s ip).

Setup the Sip ATA by Web Browser
The Sip ATA provides a built-in web server. You can use Web browser to configure the Sip ATA. First please input the IP
address in the web browser. In the end of IP address, please add the port number “:9999”. x:http://192.168.123.1:9999.
By default, NAT is on, LAN’s ip is 192.168.123.1(dial #120# to check LAN’s ip), WAN is DHCP client (dial #126# to check
WAN’s ip). You have to set your PC in the same subnet with TA in order to connect them with each other.
1.3 Login
Please input the username and password into the blank field. The default setting is:
1. For Administrator, the username is: root; and the password is: test. If you use this account to login, you can
configure all the settings.
2. For normal user, the username is: user; and the password is: test. If you use this account to login, but you can
not configure the SIP setting.
Click the “Login” button will move into the Sip ATA web based management information page.
If you change the setting in the Web Management interface, please do remember to click the “Submit” button in that page.
After you finished the change of the setting, click the “Save” function in the left side, and click the Save Button. When you
finished the setting, please click the Reboot function in the left side, and click the Reboot button in that page. After the
system restart, all the setting can work properly.
1.4 System Information
1.5 When you login, you can see the Sip ATA’s current system information like firmware version, model name etc in this
page.
Also you can see the function lists in the left side. You can use mouse to click the function you want to set up.

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1.6 Phone Book
1.7 In Phone Book function, you can setup the Phone Book. The phone book can store up to 140 groups of number.
1.7.1 In phone book, you can add/delete phone number.
1.7.1.1 If you want to add phone number in Phone Book, you need input position, name, URL (ie. Phone number). After
1.7.1.2 input, please click “Add Phone” button.
1.7.1.3 To delete a group of number, first select the number, and then click “Delete Selected” button to delete selected
number.
1.7.1.4 To delete all number, click “Delete All” button, a dialogue window will show up. Click OK button to delete all
numbers.
1.8 Phone Settings
Phone Setting contains Call Forward, SNTP Settings, Volume Settings, DND Settings, Caller ID, Flash Time Settings,
Call Waiting Settings and T.38(Fax) Settings.

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1.8.1 Call Forward: you can setup the phone number you want to forward to in this page. There are three type of Forward
mode. You can choose between All Forward, Busy Forward, and No Answer Forward by click the corresponding icon.
1.8.1.1 All Forward: All incoming call will forward to the number you choosed. You can input the name and the phone
number in Name and URL field.
1.8.1.2 Busy Forward: If you are on the phone, the new incoming call will forward to the number you choosed. You can
input the name and the phone number in Name and URL field.
1.8.1.3 No Answer Forward: If you can not answer the phone, the incoming call will forward to the number you choosed.
You can input the name and the phone number in Name and URL field. Also you have to set No Answer Fwd Time
Out for system to start to forward the call to the number you choosed.
When you finished the setting, please click the Submit button.
1.8.2 SNTP Settings: you can setup the primary and secondary SNTP Server to get the date/time information. Also you
can base on your location to set the Time Zone, and how long is needed to synchronize. When you finished the
setting, please click the Submit button.

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1.8.3 Volume Settings: you can setup the Handset Volume, PSTN-Out Volume, Handset Gain and PSTN-In Gain. When you
1.8.4 finished the setting, please click the Submit button.
1.8.4.1 Handset Volume is to set the volume for you to hear from the handset.
1.8.4.2 Handset Gain is to set the gain send out to the other side.
1.8.5 DND Setting function: you can setup the Block Setting to keep the phone slience. You can choose DND Always or
1.8.6 DND a period. When you finished the setting, please click the Submit button.
1.8.6.1 DND Always: All incoming call will be blocked until disable this feature.
1.8.6.2 DND Period: Set a time period and the phone will be blocked during the time period. If the “From” time is large
than the “To” time, the “To” time will be regarded as next day’s time.
1.8.7 Caller ID function: you can set the device to show Caller ID in your PSTN Phone or IP Phone. There are four selection
of Caller ID. You need to base on your environment to set the Caller ID function for FSK or DTMF.

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1.8.8 Dial Plan: This function provides basic dial number replacement rule.
Drop prefix: If No, when the number you dialed beginns with the number in right blank, the number in left blank will be
added automatically before your dialed number. If Yes, when the number you dialed beginns with the number in right blank,
the number will be replaced with the number in left blank.
Auto Dial Time: the number you dialed will be sent out automatically after the time you set here.
xample 1: Drop prefix: No; Replace rule 1: 002, 8613+8662
When your dialed number beginns with 8613 or 8662, 002 will be added to the beginning of your dialed number.
xample 2: Drop prefix: Yes, Replace rule 2: 006, 002+003+004+005+007+009
When your dialed number beginns with 002 or 003, 004, 005, 007, 009, 002 or 003, 004, 005, 007, 009 will be replaced with
006.
xample 3: Drop prefix: No; Replace rule 4: 007, 5xxx+35xx+21xx
When your dialed number beginns with 5 or 35, 21, and is 4 digits, 007 will be added to the beginning of your dialed
number.

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1.8.9 Flash Time Setting function: When you use a PSTN Phone, you need press the Hook to do the Flash (Switch to
another phone line or HOLD). This function is for you to set the max time you press the Hook to activate the Flash
function.
1.8.10 Call Waiting Setting function: You can nable/Disable the Call Waiting function, When you are talking with
someone, there is a new incoming call, you will hear the call waiting tone.
1.8.11 T.38(Fax) Setting: nable/disable the fax function and set fax port.
1.8.12 Hot line setting
If enabled, when you lift handset, the hot line number will be dialed out automatically.

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1.8.13 Alarm setting
If On, the phone will alarm at alarm time.
1.9 Network
In Network you can check the Network status, configure the Network Settings, DDNS settings and VLAN settings.
1.9.1 Network Status: You can check the current Network setting in this page.
1.9.2 WAN Settings
1.9.2.1 Bridge settings: nable/disable bridge mode. If you set the Bridge On, then the two Fast thernet ports will be
transparent
1.9.2.2 WAN Settings: Set network parameters for WAN. WAN can obtain IP address through 3 methods: fixed IP, DHCP
client, PPPo . You may refer to your current network environment to configure the Sip ATA properly.
1.9.2.3 The PPPo Setting: Set PPPo Username and Password. If you have the PPPo account from your Service Provider,
please input the Username and the Password correctly.
When you finished the setting, please click the Submit button.

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1.9.3 LAN Settings: Set network parameters for LAN.You may refer to your current network environment to configure the
Sip ATA properly. When you finished the setting, please click the Submit button.
1.9.4
1.9.5

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1.9.6 DDNS Setting: You can configure the DDNS setting in this page. You need to have the DDNS account and input the
informations properly. If you have a DDNS account with a public IP address, others can call you via the DDNS
account. But now most of the VoIP applications work with a SIP Proxy Server. When you finished the setting, please
click the Submit button.
1.9.7 VLAN Settings
1.9.8 DMZ setting

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1.9.9 Virtual Server
1.10 SIP Settings
In SIP Settings you can setup the Service Domain, Port Settngs, Codec Settings, Codec ID Setting, DTMF Setting, RPort
Setting and Other Settings. If the VoIP service is provided by ISP, you have to setup the related informations correctly so that
you can register to the SIP Proxy Server successfully.
1.10.1 In Service Domain Function you need to input the account and the related informations in this page. Please refer to
your ISP provider. You can register three SIP accounts in the Sip ATA. You can dial the VoIP phone to your friends via
first enabled SIP account and receive phone from these three SIP accounts.
First you have to choose in
Phone 1
to decide which phone you will setup. Then click Active to enable the Service
Domain, then you can input the following items:
Display Name: you can input the name you want to display.
User Name: you need to input the User Name got from your ISP.
Register Name: you need to input the Register Name got from your ISP.
Register Password: you need to input the Register Password got from your ISP.
Domain Server: you need to input the Domain Server got from your ISP.
Proxy Server: you need to input the Proxy Server got from your ISP.
Outbound Proxy: you need to input the Outbound Proxy got from your ISP. If your ISP does not provide the information, you
can skip this item.
Subscribe for MWI: When set to “On” a Subscribe for Message Waiting Indication will be sent periodically.
Status: You can see the Register Status in the Status item. If the item shows “Registered”, then your ATA is registered to the
ISP, you can make a phone call now.
If you have more than one SIP account, you can follow the steps to register to the other ISP.
When you finished the setting, please click the Submit button.

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1.10.2 Port Settings: you can setup the SIP and RTP port number in this page. ach ISP provider will have different
SIP/RTPport setting, please refer to the ISP to setup the port number correctly. When you finished the setting,
please click the Submit button.

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1.10.3 Codec Settings: you can setup the Codec priority, RTP packet length, and VAD function in this page. You need to
follow the ISP suggestion to setup these items. When you finished the setting, please click the Submit button.
1.10.4 Codec ID Setting: Sometimes 2 VoIP devices with different Codec ID will cause some problems. If there are some
problems when you are talking with others, you may ask the other one what Codec ID he is useing, and change your
Codec ID to the same as his. When you finished the setting, please click the Submit button.

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1.10.5 DTMF Setting: Choose between 2833, Inband DTMF and Send DTMF SIP Info. Please refer to your ISP for correct
setting. When you finished the setting, please click the Submit button.
1.10.6 RPort Setting: nable/Disable RPort. To change this setting, please following your ISP’s information. When you
finished the setting, please click the Submit button.
1.10.7 Other Settings: you can setup the Hold by RFC, Voice/SIP QoS and SIP expire time in this page. To change these
settings please following your ISP’s information. The QoS setting is to set the voice packets’ priority. If you set the
value higher, the voice packets will get higher priority. But the QoS function still need to cooperate with the others
Internet devices. When you finished the setting, please click the Submit button.

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1.11 NAT Trans.
1.11.1 STUN Setting: you can setup the STUN nable/Disable and STUN Server IP address in this page. This function can
help your ATA working properly behind NAT. To change these settings please following your ISP information. When
you finished the setting, please click the Submit button.
1.12 Others
1.12.1 Auto Config: Please set these according to your need.
1.12.2 FXS Port Setting: Please set this according to your country’s standard.
1.12.3 MAC Clone Setting: nable/disable MAC Clone.

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1.12.4 Tones Settings
1.12.5 Advanced Settings
1.13 System Auth.
In System Authority you can change your login name and password. When you finished the setting, please click the Submit
button.

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1.14 Save Change
In Save Change you can save the changes you have made. If you want to make new setting into effect, You have to click the
Save button. After you click the Save button, the Sip ATA will automatically restart and the new setting will take effect.
1.15 Update
1.16 In Update you can update the IP Phone’s firmware to the latest or restore the factory setting.
1.16.1 In New Firmware function you can update new firmware via Local PC/TFTP in this page. You can ugrade the
1.16.2 firmware by the following steps:
1) Select the firmware code type, Risc or DSP code.
2) Click the “Browse” button in the right side of the File Location, or you can type the correct path and the
filename in File Location blank.
3) Select the correct file you want to download to the IP Phone then click the Update button.