Page 1
VS110
SIP
VoIP Telephone Adaptor
User Manual
V2.1h
Page 2
2
Quick Guide
Step 1: Broadband (ADSL/Cable Modem) Connections for VS110
A. Connect VS110 LAN port to ADSL NAT Router as the following connection.
B. Connect VS110 PC port to Notebook PC LAN port using a Category 5 LAN cable.
C. Connect VS110 RJ11 PHONE port to a Telephone Set.
D. Connect DC Power Adaptor. After power on, the POWER LED will be Green ON. In 5 seconds,
the PHONE LED will start flashing 5 times and be ready for configurations.
E. Pick up the phone, the PHONE LED will be lit, and you should hear a dial tone.
F. 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 VS110. After the IP announcement, please hang up.
Figure A. ADSL Connections with NAT Router for VS110
Step 2: Settings for VS110 from PC Web Browser
A. VS110 is defaulted at embedded NAT mode.
B. Press #120# from the phone to listen and check IP address (default is 192.168.123.1) for
VS110.
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.
PC
Phone
NAT Router
VS110 IP: 192.168.123.1
LAN
PC IP: 192.168.1
23.150
ADSL Modem
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E. You need to set up the following web configurations: Phone Settings, Network, SIP Settings,
NAT Settings for registration to a SIP server. 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 user manual.
Step 3: Making Point-To-Point SIP Calls
A. While the PHONE LED is flashing continuously showing a successful registration in the SIP
server.
B. Pick up the phone, and you should hear a dial tone.
C.
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.
Note: Difficulties in configuring VS110?
Please refer to the last chapter for trouble shootings.
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TABLE OF CONTENTS
1 INTRODUCTIONS....................................................................................................................................6
2 FEATURES ..............................................................................................................................................6
3 STANDARD COMPLIANCES ..................................................................................................................7
4 PACKING CONTENTS .............................................................................................................................7
5 LED INDICATORS....................................................................................................................................7
6 INSTALLATIONS & SIP CONFIGURATIONS ........................................................................................8
7 DEFAULT RESET BY TELEPHONE........................................................................................................8
8 CONFIGURATIONS BY WEB BROWSER............................................................................................... 9
L
OGIN THE GATEWAY
................................................................................................................................ 9
S
YSTEM INFORMATION
............................................................................................................................ 10
P
HONE BOOK SETTINGS
........................................................................................................................... 11
Phone Book...................................................................................................................................... 11
P
HONE SETTINGS
..................................................................................................................................... 12
Call Forward.................................................................................................................................... 12
SNTP............................................................................................................................................... 13
Volume .......................................................................................................................................... 14
DND................................................................................................................................................ 15
Caller ID.......................................................................................................................................... 16
Dial Plan.......................................................................................................................................... 17
Flash Time....................................................................................................................................... 19
Call Waiting..................................................................................................................................... 20
T.38 (FAX) .................................................................................................................................... 21
Hot Line........................................................................................................................................... 22
Alarm............................................................................................................................................... 23
N
ETWORK
............................................................................................................................................... 24
Network Status ............................................................................................................................... 24
WAN ............................................................................................................................................. 25
LAN ............................................................................................................................................... 26
DDNS ............................................................................................................................................ 27
VLAN ............................................................................................................................................ 28
DMZ................................................................................................................................................ 30
Virtual Server ................................................................................................................................. 31
L2TP................................................................................................................................................ 32
PPTP................................................................................................................................................ 33
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SIP S
ETTINGS
.......................................................................................................................................... 34
Service Domain................................................................................................................................ 34
Port Settings ................................................................................................................................... 36
Codec Settings ............................................................................................................................... 37
Codec ID Settings: ......................................................................................................................... 38
DTMF Settings: ............................................................................................................................. 39
Rport Settings: ............................................................................................................................... 39
Other Settings .................................................................................................................................. 40
NAT T
RANS
............................................................................................................................................ 41
STUN ............................................................................................................................................ 41
O
THERS
................................................................................................................................................. 42
Auto Config..................................................................................................................................... 42
FXS Port.......................................................................................................................................... 44
MAC Clone...................................................................................................................................... 45
Tone ................................................................................................................................................ 45
Advanced......................................................................................................................................... 46
Status Log........................................................................................................................................ 47
U
SER PASSWORD
..................................................................................................................................... 48
S
AVE CHANGES
....................................................................................................................................... 48
U
PDATE
................................................................................................................................................... 49
New Firmware ............................................................................................................................... 49
Auto Update: .................................................................................................................................. 52
Default Settings: ............................................................................................................................. 56
R
EBOOT
................................................................................................................................................... 57
9 CONFIGURATIONS BY TELEPHONE & IVR....................................................................................... 58
10 VOIP APPLICATIONS EXAMPLES..................................................................................................... 60
E
XAMPLE 1:
SIP-TO-SIP C
ALLING/ANSWERING
................................................................................... 61
E
XAMPLE 2:
SIP
TO DIRECT IP CALLING
.............................................................................................. 61
E
XAMPLE 3: DIRECT IP TO DIRECT IP CALLING/ANSWERING
................................................................. 62
E
XAMPLE 4: DIRECT IP TO DIRECT IP CALLING WITHIN
NAT R
OUTER
.................................................. 62
E
XAMPLE 5:
3-W
AY CONFERENCE CALL, CALL TRANSFER, CALL WAITING, HOLD
.............................. 63
E
XAMPLE 6:
SIP-TO-SIP C
ALLING FOR HTTP
://
WWW.INPHONEX.COM
/................................................... 64
11 TROUBLE SHOOTING FOR WEB CONFIGURATIONS..................................................................... 68
11.1.
DO
NOT HEAR DIAL TONE?.................................................................................................... 68
11.2.
CAN NOT ACCESS WEB PAGE? .............................................................................................. 68
11.3.
ONLY ONE IP AVAILABLE FROM ADSL/CABLE SERVICE PROVIDER? .......................... 69
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1 Introductions
The VS110 is a single port Telephone Adaptor (TA) with SIP Protocols for Voice over IP (VoIP)
applications. Connecting to the Internet with a plain old telephone set (POTS), the VS110 can
make a voice phone call over the Internet from one IP to another. VS110 provides two Ethernet
ports for connections to Internet as well as Notebook PC. With an embedded NAT/DHCP
server, the VS110 can be easily configured by the Web browser, and is very suitable for ITSP
(Internet Telephony Service Providers) customers and SOHO users to make VoIP calls.
Note that VS110 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.
VS110 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 VS110 VoIP TA is equipped with one RJ11 connector for POTS, and two RJ45 connectors
for ADSL Modem/Router and PC connections. The VS110 is featuring as the following
Three LED Indicators for VS110: POWER, PHONE, LAN
RJ45 x 2 for WAN and LAN ports + RJ11 x 1 for FXS port
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
Support registrations for up to 3 SIP Servers
Hot Line Mode
Dial Plan Settings
T.38 FAX over IP
Interactive Voice Recording (IVR) for telephone IP status
Phone Book, Call Forward/Waiting, Call Transfer/Hold, and 3-Way Conference Calls
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, and Flash
Out-Band DTMF (RFC 2833) / In-Band DTMF / Send DTMF SIP Info
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3 Standard Compliances
The VS110 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:
⑴. One VS110 SIP TA
⑴. One AC to 12VDC/1A Power Adaptor
⑴. 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 VS110, there are three LED indicators as the following
POWER: “On” indicates the power is normal
PHONE: “On” indicates the Phone is in use (off-hook) for VoIP call.
“Flashing” indicates successful registration at SIP server.
LAN: “On” indicates the LAN port of VS110 is in Connection.
“Flashing” indicates the data activity of LAN port.
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6 Installations & SIP Configurations
1. Connect VS110 RJ45 LAN port to ADSL Modem/Router using a Category 5 LAN cable.
2. Connect VS110 RJ45 PC port to Notebook PC using a Category 5 LAN cable.
3. Connect VS110 RJ11 PHONE port to a Plain Old Telephone Set (POTS).
4. Connect the power adaptor to power on VS110, and the POWER LED will be lit constantly.
5. 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 constantly ON when any one of
RJ45 ports is connected. If the PHONE LED keeps flashing, it indicates that VS110 has
successfully registered in the SIP server.
6. Pick up the phone, the PHONE LED will be lit and you should hear a dial tone. If you hear a
busy tone, please check if the LAN port is connected properly.
7. Press #120# to listen and check the assigned IP address for the VS110. The default IP
address is 192.168.123.1. This IP address will be used for the Web configurations from
Notebook PC. Please refer to Section 8 for Web configurations.
8. After successful registration to the SIP server, the PHONE LED will keep continuous flashing.
Press 123456 to call the party with the number 123456 registered in the SIP server. In a
moment (5 seconds), you should hear the ring back tone, and wait for answer. Note that you
may press 123456# to dial out the number immediately. Dialing without # will not dial out until
the auto dial timer (default=5 seconds) elapsed.
7 Default Reset by Telephone
VS110 provides an easy way to reset to factory defaults by using Telephone.
Pick up the phone and press #198#. The VS110 will reset back to factory defaults, and enter
into POWER ON cycle. The PHONE LED indicators will be OFF for about 5 seconds and start
flashing for 5 times. The POWER LED then will be lit constantly, and the PHONE LED will be
OFF. If the PHONE LED keeps flashing, it indicates that VS110 has successfully registered in
the SIP server.
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8 Configurations by Web Browser
Login the gateway
You may enter the IP address from PC Web browser to configure VS110. 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.
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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 VS110.
Model Name Show device Model Name.
Firmware Version Device Risc version, eg: Tue Jan 16 11:28:32 2007
Codec Version Device DSP version, eg: Wed Dec 20 17:28:06 2006
Call Status
Protocol Show device VoIP protocol.
Realm #1 Show the first registry information status
Realm #2 Show the second registry information status
Realm #2 Show the third registry information status
WAN Status
MAC address WAN port of MAC address
IP address/Netmask IP address/netmask
Gateway Default router IP address
DNS DNS IP address
LAN Status
IP address/Netmask Current LAN port IP address/netmask
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Phone Book Settings
Phone Book
8.6. You may add/delete Name (in numeric only) up to maximum 140 entries in Phone book list.
8.7. To add a phone name, you need to enter the position, the name, and the phone URL. When
you finished a new phone list, just click the “Add Phone” button.
8.8. To delete a phone name, please select the phone name then click “Delete Selected” button.
8.9. To delete all phone names, please click “Delete All” button.
8.10. After dialing a phone number, the TA will first match with the Name in the phone book. If
matched, the TA will send out the corresponding URL. If not, the dialed phone number will
be sent out.
8.11. Example 1: Name: 101, URL: 192.168.1.100
Press 101# on telephone, and the phone at 192.168.1.100 will start ringing.
8.12. Example 2: Name: 102, URL: james@sipserver.com
Press 102# on telephone, and the TA will call the URL james@sipserver.com.
8.13. Example 3: Name: 103, URL: 612345
Press 103# on telephone, and the TA will call the registered number 612345.
8.14. Example 4: No Name: 401 in phone book.
Press 401# on telephone, and the TA will call the registered number 401.
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Phone Settings
8.15. The pages are as follows; Call Forward, SNTP, Volume, DND, Caller ID, Dial Plan, Flash
Time (or hook switch), Call Waiting, T.38 FAX over IP, Hot Line and Alarm settings.
Call Forward
8.16. 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.
8.17. 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
8.18. 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.
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Volume
8.19. You can setup the Handset Volume and Handset Gain in this page. Handset Volume is to set
the volume hearing from the handset. Handset Gain is to set the volume send out to the
other side’s handset.
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DND
8.20. You can setup the DND (Do Not Disturb) setting to keep the phone silence. You can choose
either DND Always or a DND period.
8.21. DND Always: All incoming call will be blocked until this feature is disabled.
8.22. DND 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 DND time will be from Day 1 to Day 2.
8.23. After you finished the setting, please click the “Submit” button.
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Caller ID
8.24. 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
8.25. Dial plan and auto dial timer settings can be set in this page. The dial plan allows you to map
the dialing into an easy-to-remember phone number system. The auto dial timer specifies the
elapse time between the dialing digits. When Drop prefix is ON and the dialing prefix is
matched, the prefix will be dropped and replaced by the rule digits and followed by the rest of
dialing digits. When Drop prefix is OFF and the dialing prefix is matched, the rule digits will
be added before the dialing digits in accord with the settings.
8.26. Symbol Representations:
Example 1: Drop Prefix: No, Replace rule 1: 002, 8613+8862
a) Pressing 8613xxx will result in dialing out 002+8613+xxx.
b) Pressing 8862xxx will result in dialing out 002+8862+xxx.
Example 2: Drop Prefix: Yes, Replace rule 2: 006, 002+003+004+005+007+009
a) Pressing 002xxx will result in dialing out 006+xxx.
b) Pressing 003xxxx will result in dialing out 006+xxxx.
Example 3: Drop Prefix: No, Replace rule 3: 009, 12
a) Pressing 12xxx will result in dialing out 009+12xxx.
Example 4: Drop Prefix: No, Replace rule 4: 007, 5xxx+35xx+21xx
a) Pressing 5xxx will result in dialing out 007+5xxx.
b) Pressing 534 will result in dialing out 534 (not matched for the rest 3 digits).
c) Pressing 35xx will result in dialing out 007+35xx.
d) Pressing 356 will result in dialing out 356 (not matched for the rest 2 digits).
e) Pressing 35668 will result in dialing out 35668 (not matched for the rest 2 digits).
Example 5: Dial Now: *xx+#xx+11x+xxxxxxxx
a) Pressing *00, *01, *02 .. *99 will result in dialing out the same *xx immediately.
b) Pressing #00, #01, #02 .. #99 will result in dialing out the same #xx immediately.
c) Pressing 110, 111, .. 119 will result in dialing out the same 11x immediately.
d) Pressing 12345678 (8 digits) will result in dialing out 12345678 immediately. This implies
that the phone numbers with 9 or more digits are prohibited.
Symbol Representations
x or X 0,1,2,3,4,5,6,7,8,9
+ or
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8.27. Auto Dial Timer: The inter-digit timer. Default is 5 seconds.
8.28. None SIP Server Mode : When SIP Settings > Service Domain were left blank (that means
no SIP registration available), you may enter a specific IP address (e.g. SIP gateway IP
address) in this field for all dialing numbers sending to this IP address. This can be used
when only SIP gateway is available. Note when at least one SIP server is successfully
registered in SIP settings, this will be automatically disabled.
8.29. When you finish the setting, please click the Submit button.
8.30. Click the Save button. The changes you have made will be saved and the VS110 will reboot
automatically.
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Flash Time
8.31. 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.
Notes:
Flash Signal Detect (MAX): Maximum Flash Hook duration (unit:10ms)
Flash Signal Detect (MIN): Minimum Flash Hook duration (unit:10ms)
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Call Waiting
8.32. 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.
Call Transfer function
The call transfer function allows users to answer an incoming call and to hold the current call by
pressing flash key, and then transfer the current call to the desired party by dialing the desired
party number ended with # key. The call transfer function is exclusive with call waiting function.
You may enable call transfer function by disabling the call waiting function (#139#), or disable call
transfer function by enabling the call waiting function (#138#).
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T.38 (FAX)
8.33. 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. T.38
Pass through codec for uLaw or aLaw, and make sure your SIP server/gateway also supports
this T.38 function. When you finished the setting, please click the Submit button.
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Hot Line
8.34. The Hot Line mode allows to making a direct call at the Phone Number or IP stored in this
page without dialing. Hot-Line Mode is very convenient for IP calling to Public Switching
Telephone Network (PSTN) number through FXO Gateway
8.35. When the Hot Line mode is enabled, you just pick up the phone and the VS110 will call the
party directly to the preset IP (or URL) address. The default for Hot Line mode is disabled.
8.36. You need to “Enable” and click the “Submit” button and reboot to activate the function.
Pick up the phone. In 1-2 seconds (default wait time), the VS110 will automatically call the preset
IP or phone number.
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Alarm
8.37. You can configure the Alarm setting in this page.
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Network
8.38. VS110 is equipped with an embedded NAT router between LAN and PC ports to meet the IP
Network requirements. If you have an external NAT router, then you may select Bridge
mode in WAN setting. The WAN setting is for the LAN port, and LAN setting for PC port. Thus
the two LAN and PC Ethernet ports will be bridged and transparent. Otherwise, you may
select NAT mode to enable embedded NAT and go on DDNS settings. The default is at NAT
mode.
Network Status
8.39. You can check and show the current Network settings in this page.
- Interface 0 is for WAN port Status and Interface 1 is for LAN port Status .
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WAN
8.40. The WAN setting is used to configure the VS110 LAN port connecting to the ADSL
Modem/Router.
8.41. The default setting is NAT mode for VS110, and this enables the embedded NAT router
between the LAN port and PC port. You may select Bridge Mode if you need NOT use the
embedded NAT router. When setting to Bridge Mode, only the WAN settings will get
effective and the LAN settings in the next section will be ignored.
8.42. There are three selections for WAN IP Type: Fixed IP, DHCP Client, and PPPoE modes. This
WAN setting is for the VS110 LAN port when set in NAT mode. The WAN default is at
DHCP Client Mode.
8.43. For Fix IP Mode, please make sure the IP address. Net Mask, Gateway, and DNS settings
are suitable in your current network environment.
8.44. For PPPoE Mode, you have to enter correct username and password to get the IP address
from your Internet Service Provider.
8.45. When you finished the settings, please click the Submit button.
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LAN
8.46. The default IP address is 192.168.123.1 for VS110, with Net Mask 255.255.255.0., and
DHCP Server enabled. The IP addresses for DHCP are from 150 to 200.
8.47. Connect your PC to the PC port, set your PC as DHCP Client mode, and the PC will get an
IP address from the VS110 automatically.
8.48. When you finished the settings, please click the Submit button.
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DDNS
8.49. You need to have a DDNS account before configuring the DDNS setting. Usually, most of the
VoIP applications are working with a SIP Proxy Server. Nonetheless, you may have a DDNS
account with a public IP address, and others can call you via the DDNS account. When you
finished the setting, please click the Submit button.
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Example :
VLAN
8.50. The VLAN setting is for VoIP packets related to LAN port.
8.51. VLAN Packets: If you enable VLAN Packets and set the VID, User Priority, and CFI, then all
the incoming packets will be checked with the IP Address and the VID.
8.52. VID: Please set your VID in accordance with your service provider.
8.53. User Priority: Defines user priority with eight (2^3) priority levels. IEEE 802.1P defines the
operation for these 3 user priority bits. Usually, this will be defined by your service provider.
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8.54. CFI: Canonical Format Indicator is always set to zero for Ethernet switches. CFI is used for
compatibility between Ethernet type network and Token Ring type network. If a frame
received at an Ethernet port has a CFI set to 1, then that frame should not be forwarded as it
is to an untagged port.
8.55. When you enable the first VLAN Packets and set the VID, User Priority, and CFI, then all the
incoming packets with the TA’s IP address and the same VID will be accept by the TA. If the
incoming packets with TA’s IP address but different VID then the packets will be discard by
the TA. The Other incoming packets with different IP address will go through the LAN port to
the PC port.
Notes :
When WAN port set to NAT Mode with VLAN enabled, please make sure the PC or
network device on LAN port must support the same VLAN function to pass through
WAN port.
When WAN port set to Bridge Mode with VLAN enabled, the VLAN will only
functionon on ATA. The packets from LAN port, will pass through WAN without
VLAN function.
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DMZ
8.56. The DMZ can be enabled/disabled and configured in this page.
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Virtual Server
8.57. The Virtual Server IP and Port numbers can be configured in this page.
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L2TP
8.58. The Layer Two Tunneling Protocol (L2TP)Server can be set ON/OFF in this page. This
L2TP can be used to pentrate the firewall when used with Virtual Private Network (VPN)
applications.
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PPTP
8.59. The Point-to-Point Tunnel Protocol (PPTP) Server can be set ON/OFF in this page. This
PPTP can be used to pentrate the firewall when used with Virtual Private Network (VPN)
applications.
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SIP Settings
8.60. You can setup the Service Domain, Port Settngs, Codec Settings, RTP Setting, RPort
Setting and Other Settings for SIP Proxy Server registrations in this page.
Service Domain
8.61. You may register up to three SIP Servers for three Realms in the VS110. You can receive the
incoming calls from all the three SIP Servers. For outgoing calls, you may select the
registration SIP server first, and then call the associated registration phone number.
8.62. To select SIP Server 1 (default), please pickup the phone, press 1*, then hangup.
8.63. To select SIP Server 2 (or 3), please pickup the phone, press 2* (or 3*), then hangup.
8.64. Click “Active” ON to enable the Service Domain, then enter the following items:
8.65. Display Name: enter the name you want to display.
8.66. User Name: enter the User Name given by your ITSP.
8.67. Register Name: enter the Register Name given by your ITSP.
8.68. Register Password: enter the Register Password given by your ITSP.
8.69. Domain Server: enter the Domain Server given by your ITSP.
8.70. Proxy Server: enter the Proxy Server given by your ITSP.
8.71. Outbound Proxy: enter the Outbound Proxy of ITSP. You may skip If not provided.
8.72. Register Period: enter the Register Period in minute given by your ITSP.
8.73. When it shows “Registered” in the Register Status, it indicates a successful registration to
the ITSP, and the “PHONE” LED will start flashing. The VS110 is then ready for VoIP call.
8.74. If you have more than one SIP account, please follow the steps to register to other ITSPs.
8.75. After you finished the setting, please click the “Submit” button.
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Click selection?
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Port Settings
8.76. You can setup the SIP and RTP port number in this page. Each ITSP provider might have
different SIP/RTP port setting, please refer to the ITSP to setup the port number correctly.
When you finished the setting, please click the “Submit” button. The defaults for SIP port
and RTP port are 5060 and 20000, respectively.
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Codec Settings
8.77. You can setup the Codec priority, RTP packet length, and VAD function in this page. You
need to follow the ITSP recommendations to setup these items.
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Codec ID Settings:
8.78. You can setup the Codec ID in this page. You need to follow the ITSP suggestion to setup
these items.
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DTMF Settings:
8.79. You can setup the options for DTMF function in this page. The options include RFC2833
(Outband DTMF), Inband DTMF, and Send DTMF SIP info. The default is set at Inband
DTMF. If you are making two-stage callings for extension to PSTN, you might need to select
Outband DTMF option.
Rport Settings:
8.80. You can enable/disable the RPort in this page. To change this setting, please follow your ISP
information. When you finished the setting, please click the Submit button.
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Other Settings
8.81. You can setup the Hold by RFC and QoS in this page. To change these settings please
follows your ITSP information. When you finished the setting, please click the Submit button.
The QoS is used to set the voice packet priority. Higher value other than zero will get higher
priority for the voice packets in Internet. However, the QoS function still needs to cooperate
with the other Internet devices.
SIP Expire Time: 60 (15~86400 sec, 0=defined by Server).
Keep Alives Period: 60 (Default: 60, Range: 15 ~ 250sec) that is send keepalives
messages in the RTP stream to keep NAT open.
Jitter Buffer: 1 (Default: 1, Range: 0~250 packets)
SIP Server Type: General, Asterisk, BroadWorks, Nortel, Xener, Vodtel.
SIP VID: 0 (2~4094, 0: disabled). This VLAN ID is for SIP only through WAN.
RTP VID: 0 (2~4094, 0: disabled). This VLAN ID is for RTP voice packets.
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NAT Trans
STUN
8.82. The STUN function must be enabled to work properly behind NAT when registered in SIP
server. You may enter the STUN server IP address and the STUN port number as shown in
the following example.
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Others
8.83. You can setup Auto Config, MAC Clone, Tone and Some Advanced Settings in this page.
Auto Config
8.84. Auto Configuration function can be used to download the original configurations stored in the
TFTP, HTTP or FTP server. This is useful for the new user to automatically download a
predefined configuration setting. Remember to click the “Submit” button and “Save” in the
Save Change section. The VS110 will then reboot and automatically download the original
configurations from the TFTP or FTP server. Note that the TFTP download works only for
public IP address.
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TFTP Mode
FTP Mode
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HTTP Mode
FXS Port
8.85. You could select the FXS impedence of the analog telephone by different country.
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MAC Clone
8.86. The MAC Clone function is to clone the MAC when only one MAC is available from ITSP.
This is to share with the PC using the same MAC. When you finished settings, please click
the Submit button.
Tone
8.87. The Tone setting can be adjusted to generate Dial tone, Ring back tone, Ring tone,
Congestion tone, Call waitting tone and Busy tone for different countries. When you finished
with the settings, please click the Submit button.
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Advanced
8.88. The advanced settings might be useful for some network requirements. The ICMP function is
to echo when someone ping this device. This can prevent from haker attacking the device by
not echoing. When you finished the setting, please click the Submit button.
IP Dialing format (P2P dialing): There are 3 options.
Type 1(xx@x.x.x), by default when you dialed 192.168.1.100 , VS110 will auto prefix
userip@ in your dialed IP address and send out to remote end.
Type 2 (x.x.x.), when you dialed 192.168.1.100 , it will send 192.168.1.100 out to
remote end without appending anything.
Disable: disable IP dialing function.
Encryption Type: Must be disabled for standard SIP protocol. Please consult your
ITSP for encryption type when registered in ITSP.
Encryption Key: Special key for encryption need
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Status Log
8.89. You can check system status log .
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User Password
8.90. You may change the login name and password in this page.
Save Changes
8.91. You can save the changes you have made, and click the Save button. After clicking the
“Save” button, the VS110 will automatically save the new settings.
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Update
New Firmware
8.92. VS110 provides two methods, HTTP or TFTP, to update new firmware as the following steps:
HTTP is mostly used for firmware update by using local PC.
TFTP is used with TFTP server for firmwares in TFTP server.
Select the firmware code type, Risc or DSP code. (mostly for Risc code)
Click the “Browse” button to choose the updated file location for HTTP download, or
Select TFTP and enter the IP address of TFTP server for firmware download, then click the
“Update” button.
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TFTP Mode
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NOTE: Do NOT power OFF the VS110 after clicking the “Select” button, or you may
damage the VS110. The remote TFTP download works only for public IP address.
8.93. After clicking the “Update” button, the firmware list will be displayed from server to indicate
the available firmware for download.
8.94. Select the new file you want to download to the VS110 then click the “Select” button.
In 3 to 4 minutes, the PHONE LED indicators will start flashing 5 times to indicate successful
firmware update. Then, you need to login again new IP address which is available from IVR
by pressing #120# from phone.
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Auto Update:
8.95. Auto update function can be used to auto-update the firmware stored in the TFTP, HTTP or
FTP server per the schedule as the following settings.
8.96. Select the update method and enter the server IP address,
8.97. Click the “Submit” button to get auto-update effective.
Notes:
Check new Firmware:
Power on and Scheduling: Check new firmware when power on and in accord with schedule.
Must be updated manually.
Scheduling Only (default): Check in accord with schedule.
Scheduling (Date): Default at 14 days.
Scheduling (Time): Default at AM 00:00 – 05:59 randomly.
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TFTP Mode
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FTP Mode
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HTTP Mode
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Default Settings:
8.98. You can restore the VS110 to factory default in this page. By clicking the “Restore” button,
the VS110 will restore to default and automatically restart again.
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Reboot
8.99. You may click the Reboot button to restart, then VS110 will automatically reboot with the
stored configurations.
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9 Configurations by Telephone & IVR
You can use telephone to configure and check the status of VS110. Note the WAN setting is for
LAN port of VS110, and the LAN setting is for PC port of VS110.
Group IVR Action Phone Command Remarks
Setting Set WAN DHCP client #111# This set WAN to DHCP Client mode.
Setting Set WAN Static IP Address #112xxx*xxx*xxx*xxx#
Ex: #112061*066*159*009#
Note xxx must be 3 decimal digits.
This setting will disable DHCP Client.
Setting Set WAN Network Mask #113xxx*xxx*xxx*xxx#
Ex: #113255*255*255*000#
Note xxx must be 3 decimal digits and
Must set WAN Static IP first (#112).
Setting Set Router IP Address #114xxx*xxx*xxx*xxx#
Ex: #114061*066*159*254#
Note xxx must be 3 decimal digits and
Must set WAN Static IP first (#112).
Setting Set Primary DNS Server #115xxx*xxx*xxx*xxx#
Ex: #115159*168*001*001#
Note xxx must be 3 decimal digits and
Must set WAN Static IP first (#112).
Status Check TA LAN IP Address #120#
IVR will report the current TA local
IP
address. Hang up while hearing end tone.
Status Check IP Type #121#
IVR will report if WAN
DHCP in enabled or
disabled. Hang up while hearing end tone.
Status Check Phone Number #122# IVR will report registered phone number.
Status Check Network Mask #123# IVR will report WAN network mask.
Status Check Gateway IP Address #124# IVR will report WAN Gateway IP address.
Status
Check Primary DNS Server
IP Address
#125#
IVR will report WAN Primary DNS
Server IP
address.
Status Check TA WAN IP Address #126# IVR will report the TA WAN IP address.
Status Check Firmware Version #128# IVR will announce the firmware version.
Status Set WAN interfaces Speed #129+[0-4]#
0: Auto, 1: 100M Full, 2: 100M Half,
3: 10M Full, 4: 10M Half
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.
Setting Set Handset Gain #131+[00~15]# Ex: #13107# and default is 06
Setting Set Handset Volume #132+[00~12]# Ex: #13209# and Default is 10
Setting
Set TFTP Server
IP Address
#135xxx*xxx*xxx*xxx# Ex: #135061*066*159*254#
Setting
Set FTP Server
IP Address
#136xxx*xxx*xxx*xxx# Ex: #136159*168*001*001#
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Group IVR Action Phone Command Remarks
Setting Enable Call Waiting #138# This will disable Call Transfer.
Setting Disable Call Waiting #139# This will enable Call Transfer.
Setting Unlock Keypad #190#
You must unlock keypad first in order to
change settings by keypad.
Setting Lock Keypad #191# Keypad can NOT be used for setting.
Setting Reboot #195#
After you hear “Option Successful,” hang-
up
and TA will reboot automatically.
Setting Factory Reset #198#
TA will reset back to factory defaults.
WARNING: ALL “User-Changeable”
NONDEFAULT SETTINGS WILL BE LOST!
Setting Blind Call Transfer #510#xxxxxx# Ex: #510#54321# transfer to 54321
Setting Attendant Call Transfer #511#xxxxxx# Ex: #511#54321# transfer to 54321
Setting 3-Way Conference Call #512#xxxxxx# Ex: #512#54321# conference with 54321
Setting 1* Select the First SIP server.
Setting 2* Select the Second SIP server.
Setting 3* Select the Third SIP server.
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10 VoIP Applications Examples
You can use PC Web browser to configure VS110.
For example, enter http://192.168.123.1 from PC web browser.
A. ADSL Connections without NAT Router for VS110
B. ADSL Connections with NAT Router for VS110
PC
P
HONE
NAT Router
ADSL Modem
VS110 IP: 192.168.123.1
LAN
PC
PHONE
ADSL Modem
LAN
INTERNET
INTERNET
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Example 1: SIP-to-SIP Calling/Answering
Applications:
The applications can be for ADSL connections as in either Diagrams A or B. Both parties
are registered to SIP server with private IP under NAT router. The SIP-to-SIP calling works
when both calling and answering parties are registered to SIP server with given registered
phone numbers. For Diagram A without NAT router, you may select NAT mode to enable
the embedded NAT router. For Diagram B with external NAT router, you may select Bridge
mode to disable the embedded NAT.
Configurations:
1.
Select either “NAT“ or “Bridge” in accord with your network in “WAN settings” page,
2.
Select “DHCP Client” to automatically get an IP address from NAT router.
3.
Remember to click the “Submit” button,
4.
Select Active “ON” in the “SIP settings / Service Domain” page,
5.
Enter the items of Register Name, Password, Proxy Server, and Outbound Proxy,
6.
Select “ON” in the “STUN setting”, if Outbound Proxy is NOT available.
7.
Upon successful SIP registration, the PHONE LED will start Green flashing.
Callings:
8.
Pick up the phone, and you should hear a dial tone for VoIP mode.
9.
Press 1688# or 1688 to call the party with the registered SIP phone number 1688. Note
that # key will dial out the number immediately. Dialing without # will not dial out until the
auto dial timer (default=5 seconds) elapsed.
Example 2: SIP to Direct IP Calling
Applications:
The application is for the calling party with ADSL connection as in either Diagrams A or B.
The calling party is registered to SIP server with either fixed real IP or private IP under NAT
router. The answering party is with fixed real IP.
Configurations:
1.
Same as in Example 1.
2.
Select “ON” in the “SIP settings/STUN setting” page without Outbound Proxy.
3.
Make sure the PHONE LED is flashing continuously with a successful SIP registration.
Callings:
4.
Pick up the phone for VoIP mode.
5.
Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of
211.21.191.4. In a moment, you should hear a ring back tone, and wait for the VoIP
called party to answer.
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Example 3: Direct IP to Direct IP Calling/Answering
Applications:
The applications are for ADSL connection without NAT router as in Diagram A. Both parties
are with fixed real IP. The Direct IP calling works when both calling and answering parties
are with known fixed IP. SIP server registrations are not required in this application.
Configurations:
1.
Select “Fixed IP”, and bridge “ON” in the “Network / WAN settings” page,
2.
Enter the items of IP, Subnet Mask, Gateway IP,
3.
Click the “Submit” button.
4.
Make sure the SIP server is OFF (default is OFF) and PHONE LED is NOT flashing.
Callings:
5.
Pick up the phone for VoIP mode.
6.
Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of
211.21.191.4. Note that # key 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 the VoIP called party to answer.
Example 4: Direct IP to Direct IP Calling within NAT Router
Applications:
For the calling party in ADSL connection with NAT router as in Diagram B, this Direct IP
calling can work when the answering parties are with fixed private IP addresses within the
same VPN network, or with fixed real IP addresses.
Configurations:
1.
Select “Fixed IP”, and bridge “ON” in the “Network / WAN settings” page,
2.
Enter the items of IP, Subnet Mask, Gateway IP,
3.
Click the “Submit” button.
4.
Make sure the SIP server is OFF (default is OFF) and PHONE LED is NOT flashing.
Callings:
5.
Pick up the phone for VoIP mode.
6.
Press 192*168*1*51# or 192*168*1*51 to call the party with the private IP address of
192.168.1.51. Press 211*21*191*4 to call the party with the real IP address of
211.21.191.4. In a moment, you should hear a ring back tone, and wait for the called
party to answer.
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Example 5: 3-Way Conference Call, Call Transfer, Call Waiting, Hold
3-Way Conference Call, Call Transfer Applications:
These are for call transfer and conferencing among Parties A, B, and C. Three parties are
registered to SIP server with either fixed real IP or private IP. There are two kinds of call
transfer; Blind Transfer and Attendant Transfer.
Blind Transfer:
Party A calls Party B. While in conversation, Party B may press Flash key to hold the call and
then press #510# [Party C number] # to transfer to Party C.
Attendant Transfer:
Party A calls Party B. While in conversation, Party B may press Flash key to hold the call and
then press #511# [Party C number] # to call and talk to Party C. Hang up from Party B, then
Party A will transfer and connect to Party C.
3-Way Conference Call:
Party A calls Party B. While in conversation, Party B may press Flash key to hold the call and
then press #512# [Party C number] # to call and talk to Party C. Press Flash key from Party
B, then Party C will join for the conferenc call.
Call Waiting Application:
When a new call is coming while you are talking, you will hear an interrupt tone and you can
push the Flash key to switch to answer the new call. You may push the Flash key to switch
between the two calls.
Call Hold Application:
You may press the Hold (Flash) key to hold the current call for a while, then press Hold key
again to resume talking.
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Example 6: SIP-to-SIP Calling for http://www.inphonex.com/
Applications:
This shows how to use inphonex as an example for free ITSP provider. The applications are
for both parties registered to inphonex SIP server.
1. Visit http://www.inphonex.com/ and click “Test our service with free internet calling!” ICON
to sign up for a new registered account number. Follow the instructions for registration.
2. After finished, you will receive a mail sent by the inphonex mail system, and you will get
one inphonex phone number and password in the mail. For example, the register
name/phone number is 7123456 with password xxxx.
3. Login to the Web configuration page.
Configurations:
4. WAN (for VS110 LAN port) and LAN (for VS110 PC port) Settings
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5. SIP Settings
You have to enter the Display Name, User Name, Registered Name, Registered Password,
Domain Server (sip.inphonex.com), Proxy Server (sip.inphonex.com), Outbound Proxy
(keep to blank). After finished the setting, click the Submit button and the Save Change
button. The system will reboot automatically. After the system boots up, the SIP setting
page will show “Registered”, and the PHONE LED will start flashing.
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INPHONEX SIP Server Register Name: 7123456
Password: xxxx
Domain Server: sip.inphonex.com
Proxy Server: sip.inphonex.com
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Callings:
6. Pick up the phone for VoIP mode. (Your INPHONEX phone number 7123456).
7. Press 7123455 to call the party with the registered INPHONEX phone number 7123455.
In a moment, you should hear the ring back tone, and wait for the called party to answer.
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11 Trouble Shooting for Web Configurations
11.1. DO NOT HEAR DIAL TONE?
When you pick up the phone and hear a busy tone, it indicates the VS110 LAN port is NOT
connected or failed registration at SIP Server. Make sure the ADSL Ethernet cable is connected
to the LAN port of VS110 and Power Reset again.
11.2. CAN NOT ACCESS WEB PAGE?
IE Web Browser is a useful tool to configure VS110. When you have difficulties in accessing the
default IP address http://192.168.123.1 of VS110 as in the following figure, the most possibility is
that the PC might have different subnet IP settings from 192.168.123.xxx. In this case, you must
change VS110 IP address to the same subnet as PC and NAT router.
Example: To change VS110 IP address to the same subnet as PC and NAT router
1. Pick up the phone and press #111# from the phone to enable DHCP Client mode. VS110 will
reboot, and LED will start flasing to get an IP address from NAT DHCP server.
2. Pick up the phone and press
#126#
to obtain the VS110 IP address from telephone IVR, for
example, 192.168.62.51.
3. Enter from IE web browser http://192.168.62.51 to login VS110 web page for configurations.
PC
PHONE
NAT Router
LAN
INTERNE
T
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11.3. ONLY ONE IP AVAILABLE FROM ADSL/CABLE SERVICE PROVIDER?
Sometimes only one IP address is available from Internet Service Provider (ISP) without NAT
router as the following figure. Usually, a DHCP or PPPoE server at the central side of ISP is used
to assign one IP address to each user. In this case, you may need to enable the embedded NAT
router of VS110 to provide more than one IP addresses for PC and VS110.
Example: To change PC IP address to the same subnet as 192.168.123.1 for VS110
1. As in Window 2000 (my computer),
- At "Network and Dialup Connections", right click on "Local Area Connection", then click on
property.
- The "Local Area Connection Properties" window will pop up.
- Double click on "Internet Protocol (TCP/IP)".
- The "Internet Protocol (TCP/IP) Properties" window will pop up.
- Click on "Use the following IP Address".
- Enter IP: 192.168.123.50 (50 can be any number other than 1, which is used by VS110).
- Enter Subnet mask: 255.255.255.0
- Enter Default gateway: 192.168.123.1
- Click on OK button.
2. You will lose internet connection at this time.
3. At IE browser, type http://192.168.123.1
4. Follow the example in "Advanced Settings for Embedded NAT" for web login.
5. At LAN setting, turn on DHCP server.
6. At WAN setting, choose "DHCP client" to work with your ADSL/Cable modem.
7. Save change, wait for VS110 to reboot.
8. Change your PC's "Internet Protocol (TCP/IP) Properties" back to "obtain an IP address
automatically".
9. You may press #120# and #126# to listen to the IVR for LAN and WAN IP addresses.
PC
PHONE
ADSL Modem
LAN
INTERNET
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