VS211 1 FXS / 1 FXO VoIP Telephone Adaptor User Manual V1.
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.
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.
TABLE OF CONTENTS 1. Introductions……………………………………………………………… 1 2. Features ……………………………………………………………………… 1 3. Standard Compliances…………………………………………… 2 4. Packing Contents …………………………………………………… 2 5. LED Indicators…………………………………………………………… 2 6. Installations & SIP Configurations …………………… 3 7. Default Reset by Telephone ………………………………… 3 8. Configurations by Web Browser………………………… 4 9. Configurations by Telephone & IVR…………………… 33 10.
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).
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.
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.
. 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.
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.
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.
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.
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.
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.
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.
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.
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.
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: Symbol x or X + Representations 0,1,2,3,4,5,6,7,8,9 or 8.29.
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.
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.
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.
Network 8.41. VS211 is equipped with an embedded NAT router between WAN and LAN ports to meet the IP Network requirements. If you have an external NAT router, then you may select Bridge mode in WAN setting. Thus the two WAN and LAN 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 for NAT mode. Network Status: 8.42. You can check and show the current Network settings in this page.
WAN Settings: 8.43. The WAN setting is used to configure the Ethernet port connects to the ADSL Modem/Router. 8.44. The default setting is for NAT mode to enable the embedded NAT router between the WAN port and LAN 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 will be ignored. 8.45. There are three selections for WAN IP Type: Fixed IP, DHCP Client, and PPPoE modes.
LAN Settings: 8.49. The default IP address is 192.168.123.1, with Net Mask 255.255.255.0., and DHCP Server enabled. The IP addresses for DHCP are from 150 to 200. 8.50. Connect your PC to the LAN port, set your PC as DHCP mode, then the PC will automatically get an IP address from the TA. 8.51. When you finished the settings, please click the Submit button.
DDNS Setting: 8.52. 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. VLAN Setting: 8.53. There are two parts for VLAN settings. One is to set for VoIP packets related to VS211, and the other is for the VLAN setting in the NAT Mode. 8.54.
8.58. 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. 8.59. 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.
SIP Settings 8.61. 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 Settings: 8.62. You may register up to three SIP accounts in the VS211. You can call your friends via firstly enabled SIP account and receive the phone calls from all the three SIP accounts. 8.63. Click “Active” ON to enable the Service Domain, then enter the following items: 8.64.
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Codec Settings: 8.78. 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.
Codec ID Settings: 8.79. You can setup the Codec ID in this page. You need to follow the ITSP suggestion to setup these items. Other Settings: 8.80. 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.
Auto Configuration Setting 8.81. 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 VS211 will then reboot and automatically download the original configurations from the TFTP or FTP server.
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User Password 8.82. You may change the login name and password in this page. Save Changes 8.83. You can save the changes you have made, and click the Save button. After clicking the “Save” button, the VS211 will automatically save the new settings.
Update 8.84. VS211 provides two methods, HTTP or TFTP, to update new firmware as the following steps: 8.85. Select the firmware code type, Risc or DSP code. (mostly for Risc code) 8.86. Click the “Browse” button to choose the updated file location for HTTP download, or 8.87. Select TFTP and enter the IP address of TFTP server for firmware download, then click the “Update” button.
8.88. After clicking the “Update” button, the firmware list will be displayed from server to indicate the available firmware for download. 8.89. Select the new file you want to download to the VS211 then click the “Select” button. 8.90. 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. 8.91.
Default Setting: 8.93. You can restore the VS211 to factory default in this page. By clicking the “Restore” button, the VS211 will restore to default and automatically restart again.
Reboot 8.94. You may click the Reboot button to restart, then VS211 will automatically reboot with the stored configurations.
9. Configurations by Telephone & IVR You can use telephone to configure and to check the status of VS211. Make sure that the LAN port is connected to Ethernet, or you may hear a busy tone from the telephone. Group IVR Action Phone Command Remarks IVR will report the current TA local IP address. Hang up while hearing end tone. IVR will report if WAN DHCP in enabled or disabled. Hang up while hearing end tone.
Group IVR Action Phone Command 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.
10. VoIP Application Examples You can use PC Web browser to configure VS211. For example, enter http://192.168.123.1 from PC web browser. A. ADSL Connections without NAT Router for VS211 ADSL Modem WAN LAN LINE INTERNET PSTN PHONE B. ADSL Connections with NAT Router for VS211 ADSL Modem INTERNET NAT Router WAN VS211 IP: 192.168.123.
Example 1: Public Switched Telephone Network (PSTN) Calling/Answering Applications: VS211 is default at the VoIP mode. For PSTN calls, you may just pick up the phone, press 0* key, and dial directly to the PSTN number like a normal telephone. Configurations: 1. Select “ON” for the “Auto Answer” in Call settings. 2. Select a value for Auto Answer Ring Counter, and the default value is set at 3. 3. Click the “Submit” button. Calling/Answering 4.
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 3: SIP to PSTN Calling Applications: The applications can be for ADSL connections as in both Diagrams A and B.
Example 4: 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 2. 2. Select “ON” in the “SIP settings / STUN setting” page, if Outbound Proxy is NOT available. 3.
Example 6: Direct IP to Direct IP Calling/Answering Applications: The applications are for ADSL connection without NAT router as in Diagram A. are with fixed real IP. Both parties 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.
Example 8: Direct IP to PSTN Calling Applications: The Direct IP to PSTN calling is for the application when the answering parties are with known fixed real IP addresses. The SIP server registrations may not be necessary. Configurations: 1. Same as in Example 6. 2. Select “ON” for the “Auto Answer” and “PIN Code” in Call settings. Set the Auto Answer Ring Counter, e.g. 3, and the PIN code, e.g. 1234. Callings: 3. Pick up the phone for VoIP mode. 4.
Example 10: 3-Way Conference Call, Call Waiting, Call Hold 3-Way Conference Calling Application: This is for 3-way conference call among Parties A, B, and C. Three parties are registered to SIP server with either fixed real IP or private IP. Callings: 1. Make a phone call from Party A to the first phone number Party B. 2. After the first call is established, press Flash key (or hook switch) from Party A to hold the call, and Party A should hear a dial tone. 3.
Example 11: SIP-to-SIP Calling for FreeWorld Dialup (FWD) Applications: This shows how to use FWD as an example for free ITSP provider. The applications are for both parties registered to FWD SIP server. 1. Visit http://www.freeworlddialup.com and sign up for a new registered account number. Follow the instructions for registration. 2. After finished, you will receive a mail sent by the FWD mail system, and you will get one FWD phone number and password in the mail. number is 636346 with password xxxx.
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5. SIP Settings You have to enter the Display Name, User Name, Registered Name, Registered Password, Domain Server (fwd.pulver.com), Proxy Server (fwd.pulver.com), Outbound Proxy (fwdnat.pulver.com:5082). After finished the setting, click the Submit button and the Save Change button. The system will reboot automatically. After system boot up, the SIP setting page will show “Registered”, and the PHONE LED will start flashing. FWD SIP Server Register Name: 636346 Password: xxxx Domain Server: fwd.
Callings: 6. Pick up the phone for VoIP mode. (Your FWD phone number 636346). 7. Press 654321 to call the party with the registered FWD phone number 654321. moment, you should hear the ring back tone, and wait for the called party to answer.
11. Trouble Shooting for Web Configurations 11.1. DO NOT HEAR DIAL TONE? The phone port of VS211 is set to VoIP mode at default. When you pick up the phone and hear a busy tone, it indicates the WAN port is NOT connected. Make sure the ADSL Ethernet cable is connected to the WAN port of VS211 and Power Reset again. You may press 0* key to switch to PSTN line, and you should hear another dial tone from PSTN. If not, please make sure the PSTN line is connected to the LINE port. 11.2.
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 VS211 to provide more than one IP addresses for PC and VS211.
11.4. VOIP EXTENSION CALLS TO PSTN ARE NOT WORKING? You must enable the Auto Answer function in Call Settings in order to answer automatically by VoIP mode. The Auto Answer is disabled at default. Make sure the PSTN is connected to LINE port. When the ring count exceeds the number set in Auto Answer Counter, the FXO port will auto answer and allow for VoIP extension call.
