Ansel VS211 User manual

VS211
1 FXS / 1 FXO
VoIP Telephone Adaptor
User Manual
V1.1p

Quick Guide
Step 1: Broadband (ADSL/Cable Modem) Connections for VS211
A. Connect VS211 WAN port to ADSL NAT Router as the following connection.
B. Connect VS211 LAN port to Notebook PC LAN port using a Category 5 LAN cable.
C. Connect VS211 RJ11 PHONE port to a Telephone Set.
D. Connect VS211 RJ11 LINE port to a PSTN Telephone Line.
E. Connect 12VDC Power Adaptor. After power on, the POWER LED will be Green ON.
F. The PHONE LED will be Green flashing for a successful SIP registration. Pick up the phone,
and it will be Green ON indicating VoIP mode.
G. Press 0*key, and the PHONE LED will be Yellow ON indicating PSTN mode.
H. Press #121# and #120# from the phone to listen to IVR and to check the DHCP status and the
IP address (e.g. 192.168.123.1) for VS211. After the IP announcement, please hang up.
Figure A. ADSL Connections with NAT Router for VS211
Step 2: Settings for VS211 from PC Web Browser
A. This is an example for FWD ITSP provider using ADSL connection with NAT router as in Figure
A. Both parties sign up to FWD SIP server http://www.freeworlddialup.com with individual
registered FWD phone numbers and passwords.
B. Pick up the phone and press #120# from the phone to listen to IVR and get the IP address (e.g.
192.168.123.1) for VS211.
C. Enter the IP address from PC Web browser for configuration settings.
Example:Enter http://192.168.123.1 from IE Web browser to display login page.
D. Enter the user name and password into the blank field. The default settings are Username:
root Password: test . Click the “Login” button to enter for configurations.
i
LAN
PHONE
NAT Router
ADSL Modem
VS211 IP: 192.168.123.1
W
AN
PC IP:
192.168.1
23
.1
50
LINE
PSTN
INTERNET

E. You need to set up the following web configurations: Network Settings, SIP Settings, NAT
Settings/STUN Settings. Remember to submit, save and reboot for new configurations.
F. The PHONE LED will be Green flashing showing a successful registration in the SIP server.
For further detail configurations, please refer to the VoIP applications chapter.
Step 3: Making Point-To-Point SIP Calls
A.
Pick up the phone in VoIP mode, and you should hear a dial tone.
B.
Press 123456# to call the party with the number 123456 registered in the SIP server. Note #is
used to send out the call immediately. In a moment, you should hear the ring back tone, and
wait for the called party to answer. For more applications, please refer to the user manual
.
Step 4: Making Public Switching Telephone Calls
A.
Pick up the phone.
B. Press 0*key for PSTN mode, and the PHONE LED will be Yellow ON
, and you should hear a dial
tone
. Now you may dial any public phone number.
Note: Difficulties in configuring VS211? Please refer to the last chapter for trouble shootings.
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TABLE OF CONTENTS
1.
Introductions
………………………………………………………………
2.
Features
………………………………………………………………………
3.
Standard Compliances
……………………………………………
4.
Packing Contents
……………………………………………………
5.
LED Indicators
……………………………………………………………
6.
Installations & SIP Configurations
……………………
7.
Default Reset by Telephone
…………………………………
8.
Configurations by Web Browser
…………………………
9.
Configurations by Telephone & IVR
……………………
10.
VoIP Applications Examples
………………………………
PSTN Calling/Answering
………………………………………
SIP-to-SIP Calling/Answering
……………………………
SIP to PSTN Calling
………………………………………………
SIP to Direct IP Calling
…………………………………………
PSTN to SIP Calling
………………………………………………
Direct IP to Direct IP Calling/Answering
………
Direct IP to Direct IP Calling within NAT
…………
Direct IP to PSTN Calling
……………………………………
PSTN to Direct IP Calling
……………………………………
3-Way Conference Call, Call Waiting, Hold
……
SIP-to-SIP Calling for FWD
…………………………………
11.
Trouble Shooting for Web Configurations
……
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1. Introductions
The VS211 is a 1-port FXS / 1-port FXO Telephone Adaptor (TA) with SIP Protocols for Voice
over IP (VoIP) applications. Connecting to the Internet and the PSTN line with an analog
telephone set, the VS211 can connect a VoIP call over the Internet with extension to the public
switched telephone line. VS211 provides one WAN port for Internet connections, one LAN port
for Notebook PC, and two RJ11 connectors for Phone (FXS) and PSTN (FXO). With an
embedded NAT/DHCP server, VS211 can be easily configured for different network diagrams by
PC Web browser and telephone set. This is very suitable for ITSP (Internet Telephony Service
Providers) and SOHO users to make 2-stage VoIP with PSTN extension calls.
Note that VS211 requires an IP address, a subnet mask, and its gateway Router IP address for
its own use to connect to Internet. These three are available from your Internet service provider.
VS211 may enable PPPoE or DHCP features to automatically get an assigned dynamic IP from
the ITSP. Please refer to Section 8 Configurations by Web browser for detailed information.
2. Features
The VS211 VoIP TA is equipped with two RJ11 connectors and two RJ45 connectors and is
featuring as the following
Three LED Indicators for VS211: POWER, PHONE, LAN
RJ45 x 2 for WAN and LAN ports + RJ11 x 2 for FXS and FXO ports
Configurations by Web Browser and Telephone
Embedded NAT/DHCP Server
PPPoE/DHCP Client for Dynamic IP plus NAT, DNS, and DDNS Clients
Support STUN server for NAT Traversal
Hot Line Mode
Dial Plan Settings
T.38 FAX over IP
Interactive Voice Recording (IVR) for telephone IP status
Speed Dial, Call Forward/Waiting/Hold, and 3-Way Conference Call features
Auto Configurations by TFTP, HTTP, or FTP server
Remote Firmware Upgraded with HTTP or TFTP server by Web PC
Direct IP/URL Dial without SIP Proxy or Dial number via SIP server
Telephone features: Volume Adjustment, Phone book, Speed Dial, Redial, and Flash
Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info
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3. Standard Compliances
The VS211 VoIP TA supports for the following standards
VoIP Protocol: IETF RFC3261 and RFC 2543 for SIP
SIP Authentication: IETF RFC2069 and RFC 2617 for MD5
Speech Codec: ITU-T G.711, G.723, G.729A/B, VAD and CNG
Echo Cancellation: ITU-T G.165/168
4. Packing Contents
Inside the package you should find:
(1) One VS211 SIP TA
(2) One AC to 12VDC/1A Power Adaptor
(3) One User Manual CD
Please check if the packing is damaged or any component is missing. If so, please contact
your distributor.
5. LED Indicators
On the front panel of VS211, there are three LED indicators as the following
POWER: “On” indicates the power is normal
PHONE: “Green On” indicates the Phone is in use (off-hook) for VoIP call.
“Red On” indicates the PSTN Line is in use or ringing from incoming call.
“Yellow On” indicates the Phone and the PSTN Line are in use for PSTN call.
“Green Flashing” indicates successful registration at SIP server.
“Yellow Flashing” indicates PSTN incoming call with successful SIP registration.
LAN: “On” indicates the Ethernet port is in Connection.
“Flashing” indicates the data activity of Ethernet port.
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6. Installations & SIP Configurations
1.
Connect VS211 RJ45 WAN port to NAT Router using a Category 5 LAN cable.
2. Connect VS211 RJ45 LAN port to Notebook PC using a Category 5 LAN cable.
3. Connect VS211 RJ11 PHONE port to a Plain Old Telephone Set (POTS).
4. Connect VS211 RJ11 LINE port to a Public Switched Telephone Network (PSTN) line.
5. Connect the power adaptor (12VDC) to power on VS211, and the POWER LED will be ON.
6. The PHONE LED indicators will be OFF for about 5 seconds and start flashing for 5 times,
and remain OFF for VoIP configurations. The LAN LED will be ON when RJ45 Ethernet
ports are connected. Pick up the phone, and the PHONE LED will be Green ON indicating
VoIP mode. If you hear a busy tone, please check if the WAN port is connected.
7. Press 0* key to PSTN line. The PHONE LED will become Yellow ON and you should hear a
PSTN dial tone. If not, please check if the PSTN Line is connected.
8. Press #120# to check the assigned IP address for the VS211. The default IP address is
192.168.123.1. Please refer to Chapter 8, and you may enter this IP address in IE Web
browser for web configurations.
9. Register VS211 into your SIP server, and refer to Chapter 10 for VoIP applications examples
of SIP registrations.
10. If VS211 has successfully registered in the SIP server, the PHONE LED will start Green
flashing.
11.
Pick up the phone,
a
nd you should hear a dial tone.
Press 123456# to call the party with the
number 123456 registered in the SIP server. Note that #will dial out the number immediately.
Dialing without # will not dial out until the auto dial timer (default=5 seconds) elapsed. In a
moment, you should hear a ring back tone, and wait for answer.
7.
Default Reset by Telephone
VS211 provides an easy way to reset to factory defaults by using Telephone.
Pick up the phone and press #198# to reset back to factory defaults, and the VS211 will enter
into POWER ON cycle. The PHONE LED indicators will be OFF for about 4 seconds and start
flashing for 5 times. The POWER LED then will be lit constantly, and the PHONE LED will be
OFF.
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8. Configurations by Web Browser
You may enter the IP address from PC Web browser to configure VS211. For example, enter
http://192.168.123.1 from IE web browser to display login page as follows.
8.1. Please enter the default IP address http://192.168.123.1 from PC Web browser.
The following Web page shall be displayed on PC. If you have difficulties accessing the
Web page from the PC Web browser, the subnet IP of PC might be different from
192.168.123.xxx.In this case, please refer to Chapter 11 for trouble shooting.
8.2. Please enter the username and password into the blank field. The default settings are:
Username: root
Password: test
8.3. Click the “Login” button will enter the management information page for system setup.
Note that whenever you change the setting in each Web page, please remember to click the
“Submit” button in the page, and click the “Save” button to save into the non-volatile memory
and click the “Reboot” button to activate the new settings.
4

System Information
8.4. You will see the system information such as firmware version, Codec, etc in this page.
8.5. You may click the button list at the left hand side to configure the VS211.
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Phone Book
8.6.
The Phone Book page specifies Speed Dial and Hot Line function.
Speed Dial Settings
8.7. For Speed Dial function you can add/delete Speed Dial number up to maximum 10 entries in
Speed Dial Phone List.
8.8. If you need to add a phone number into the Speed Dial list, you need to enter the position, the
name, and the phone number (by URL type). When you finished a new phone list, just click
the “Add Phone” button.
8.9. If you want to delete a phone number, please select the phone number you want to delete
then click “Delete Selected” button.
8.10. If you want to delete all phone numbers, please click “Delete All” button.
8.11. Example: Press 2# on telephone to Speed Dial the phone number 2immediately.
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Hot Line Setting
8.12. The Hot Line mode allows to making a direct call at the IP stored in the page without dialing.
Hot-Line Mode is very convenient for IP calling to Public Switching Telephone Network
(PSTN) number through FXO Gateway
8.13. When the Hot Line mode is enabled, you just pick up the phone and the VS211 will call the
party directly to the preset IP (or URL) address. The default for Hot Line mode is OFF.
8.14. You need to set “ON” and click the “Submit” button and reboot to activate the function.
8.15. Pick up the phone. In 1 second (default wait time), the VS211 will automatically call the preset
IP number. Note that during the wait time you may press different IP phone number to make a
regular IP call, and the Hot Line function will be temporarily stopped.
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Call Settings
8.16. The subpages are as follows; Call Forward, SNTP, Volume, Block Settings, Auto Answer,
Caller ID, Dial Plan, Flash Time (or hook switch), Call Waiting, and T.38 FAX over IP.
Call Forward function:
8.17. You can select the forward mode and enter the forward URL.
All Forward: All incoming call will forward to the URL you choose.
Busy Forward: The incoming call will forward to the URL when the callee is busy.
No Answer Forward: The incoming call will forward to the URL when no answer.
8.18. You need to set the Time Out ring which will initiate No-Answer forwarding to the number you
choose. When you finished the setting, please click the “Submit” button.
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SNTP Setting:
8.19. You can setup the primary and second SNTP Server IP Address, to get the date/time
information. You may also set the Time Zone, and how long need to synchronize again.
When you finished the setting, please click the “Submit” button.
Volume Setting:
8.20. You can setup the Handset Volume/Gain, PSTN-Out Volume, and PSTN-In Gain in this page.
Handset Volume is to set the volume hearing from the handset. PSTN-Out Volume is to set
the PSTN volume for you to hear. Handset Gain is to set the volume send out to the other
side’s handset. PSTN-In Gain is to set the volume send out to the other side’s handset.
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Block Setting:
8.21. You can setup the Block Setting to keep the phone silence. You can choose either Always
Block or a Block period.
8.22. Always Block: All incoming call will be blocked until this feature is disabled.
8.23. Block Period: Set a time period and the phone will be blocked during the time period. If the
time in “From” is greater than that in “To” time, the Block time will be from Day 1 to Day 2.
8.24. After you finished the setting, please click the “Submit” button.
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Auto Answer Setting:
8.25. You may enable the Auto Answer function to answer the incoming call by VS211 FXO port.
When the ring count exceeds the number set in Auto Answer Counter, the FXO port will auto
answer and allow for extension calls from PSTN to VoIP and vice vesa. If the incoming call is
from PSTN, then VS211 FXO port will answer with a short beep tone and allow caller to redial
to VoIP number. If the incoming call is from the Internet, then FXO port will answer with a
PSTN dial tone and allow caller to redial to PSTN phone number. PIN Code is used to
prevent from call piracy. The caller needs to enter the right PIN code and the postfix “#” to get
the PSTN dial tone. Incorrect PIN Code will result in call disconnect. The Anto Answer is
disabled at default.
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Caller ID Setting:
8.26. You may show caller ID in your PSTN Phone or IP Phone by selecting “Yes” in Single Caller
ID, and the desired Caller ID option for either FSK or DTMF. When you finished the setting,
please click the “Submit” button.
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Dial Plan Settings:
8.27. The dial plan allows you to map the dialing into an easy-to-remember phone number system.
When replace prefix code is OFF, the calling number are the same as the dialing number.
When the prefix code is ON, the auto prefix rules will be applied in accord with the settings.
The auto dial timer specifies the elapse time to start sending out calling number after finished
dialing.
8.28.
Symbol Representations:
8.29.
Replace rule: If replace prefix code is ON and the prefix number is matched with rule 001 or 006 or 009
as the example web page below, then the prefix will be replaced with 005.
8.30.
Dial Plan: When the first dialing digits matched with one of the patterns, VS211 will send out the
matched corresponding number immediately.
8.31.
Example:
8.32.
Auto Prefix: Prefix number will be added if the Dial Plan or Unset Plan are NOT matched.
8.33.
Prefix Unset Plan: When first dialing digits or dial numbers match with pattern then VS211 will ignore
auto prefix.
8.34.
Example:
8.35.
Auto Dial Timer : The inter-digit timer. Default is 5 seconds.
8.36.
When you finish the setting, please click the Submit button.
8.37.
Click the Save Change Item in the left side, then click the Save button. The changes you have made
will be saved into the system and the system will Reboot automatically.
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Symbol Representations
x or X 0,1,2,3,4,5,6,7,8,9
+ or
Dial Plan Descriptions
*xx To dial one of *00, *01, …,*99 will send out the number immediately
#xx To dial one of #00, #01, …,#99 will send out the number immediately
10x To dial one of 100,101, …,109 will send out the number immediately
11x To dial one of 110,111, …,119 will send out the number immediately
Xxxxxxxx To dial 8 digits will send out the number immediately
Prefix Unset Plan Descriptions
0 Ignore auto prefix if first digit is ‘0’
1 Ignore auto prefix if first digit is ‘1’
xxxx Dialing with 4 digits will ignore auto prefix
xXxXxX Dialing with 6 digits will ignore auto prefix

Flash Time Setting:
8.38. You can set the flash time duration for the telephone flash key or hook switch in this page.
The telephone flash key is used to switch to the other phone line or HOLD, and is quite useful
for the 3-way conference call and the call waiting function. When you finished the setting,
please click the “Submit” button.
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Call Waiting Setting:
8.39. You can enable the call waiting function in this page. It allows answering another coming
call by pressing flash key while holding the current call. You may switch back to previous call
by pressing flash key again. When you finished the setting, please click the “Submit” button.
15

T.38 (FAX) Setting:
8.40. T.38 function can be used for FAX transmission over IP. Note that T.38 function must be
enabled for both side of FAX over IP. You may enable or disable the T.38 function. When you
finished the setting, please click the Submit button.
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Table of contents