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1. Auto Update Settings You could set auto update settings in this page Update via Oof OTIP OFP HTIP niese Oo TFTP File Path Exp download HTTP Server voip sipserver com Exp 60 35 187 30 PFPA et ead FTP Server Exp 60 35 17 1 Check new firmware Power ON and Scheduling Scheduling only Scheduling Time AM 00 00 05 59 vw Firmware File Prefix TA1S10 Next update time 53 Default Setting 8 98 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 Restore Default Settings You could click the restore button to restore the factory settings Phone Book Restore default settings Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Default Settings Reboot 8 99 You may click the Reboot button to restart then VS211 will automatically reboot with the stored configurations Restore Default Settings You could click the restore button to restore the factory settings Phone Book Restore default settings Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Update Reboot 54 9 Configurations by Telephone 8 IVR You can use telephone to configure and to check the status of VS211 Note that the WAN port is for WAN port of VS211 and the LAN port is for LAN port of VS211
2. Tone 8 87 The Tone setting can be adjusted to generate Dial tone Ring back tone Ring tone Congestion tone Call waiting tone and Busy tone for different countries When you finish with the settings please click the Submit button Tones Settings You could configure your tones settings in this page 5 P A Call Tone Tone Tone Tone O Cadence On Hi Tone Freq 440 480 620 620 480 440 SIP Settings Lo Tone Freq 350 440 480 480 440 350 NAT Trans Hi Tone Gain 4522 2261 2261 2261 115360 2261 Auto Config Lo Tone Gain 4522 2261 2261 2261 15360 1130 jse FXO amp FXS Port On Time 1 o 200 so 30 200 f 30 Sav MAC Clone Off Time 1 0 400 50 20 400 20 done On Time 2 0 0 0 0 0 30 Advillced Off Time 2 0 0 0 0 0 400 Reb Status Log On Time 3 0 0 0 0 0 0 Off Time 3 po o 0 o 0 0 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 hacker attacking the device by not echoing When you finish the setting please click the Submit button 1 IP Dialing format P2P dialing There are 3 options 2 Type 1 xx x x x by default when you dial 192 168 1 100 VS211 will auto prefix userip in your dialed IP address and send out to remote end 3 Type 2 x x x when you dialed 192 168 1 100 it will send 192 168 1 100 out to remote end without app
3. IVR Action Phone Command Remarks IVR will report the current TA local IP Status Check TA LAN IP Address ass Hang up while hearing end tone IVR will report if WAN DHCP in enabled or Status Check IP Type moe ieod Hang up while hearing end tone Status Check Phone Number mo IVA will report registered phone number Status Check Network Mask most IVA will report WAN network mask Status Check Gateway IP Address m2 IVR will report WAN Gateway IP address Status Check Primary DNS Server IVR will report WAN Primary DNS Server IP V IP Address address Status Check TA WAN IP Address mor 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 Se WAN DHCP client me ris set WAN to DHCP Client mode Ex 112061 066 159 009 Setting Set WAN Static IP Address Note xxx must be 3 decimal digits This setting will disable DHCP Client Ex 113255 255 255 000 Setting Set WAN Network Mask Note xxx must be 3 decimal digits and Must set WAN Static IP first 112 Ex 114061 066 159 254 Setting Set Router IP Address 114XXX XXX XXX XXX Note xxx must be 3 decimal digits and Must set WAN Static IP first 112 Ex 115159 168 001 001 Setting Set Primary DNS Server Note xxx must be 3 decimal digits and Must set WAN Static IP first 112 1 G 711 u
4. Mask Gateway IP Click the Submit button Make sure the SIP server is OFF default is OFF and PHONE LED is NOT flashing ah Callings 5 Pick up the phone for VolP 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 61 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 VolP mode 4 Press 211 21 191 4 or 211 21 191 4 to call the party with the real IP address of 211 21 191 4 After 4 rings the FXO port will auto answer with a short beep tone not dial tone Press 1234 for PIN code and then you will hear a dial tone from PSTN PIN Code is used to prevent from call piracy Incorrect PIN Code will result in call disconnect Note that if PIN code is OFF the caller may press PSTN number directly 5 Dial directly the number e g 1234567 to call the PSTN party number of 1234567 In a moment you should hear
5. No Replace rule 1 002 8613 8862 a Pressing 8613xxx will result in dialing out 0028613xxx b Pressing 8862xxx will result in dialing out 0028862xxx Example 3 Drop Prefix Yes Replace rule 2 006 002 003 004 005 007 009 a Pressing 002xxx will result in dialing out 006xxx b Pressing 003xxxx will result in dialing out 006xxxx Example 4 Drop Prefix No Replace rule 3 009 12 a Pressing 12xxx will result in dialing out 00912xxx 17 Example 5 Drop Prefix No Replace rule 4 007 5xxx 35xx 21xx a Pressing 5xxx will result in dialing out 0075xxx b 6 d e Pressing 534 will result in dialing out 534 not matched for the rest 3 digits Pressing 35xx will result in dialing out 00735xx Pressing 356 will result in dialing out 356 not matched for the rest 2 digits wor Pressing 35668 will result in dialing out 35668 not matched for the rest 2 digits Example 6 Dial Now xx xx 11xX 1 237 xx xxxXxXXXX 8 27 8 28 8 29 8 30 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 1200 1299 1300 1399 1700 1799 will result in dialing out the same 1 237 xx immediately e Pressing 12345678 8 digits will result in dialing out 12345678 immediately This implies that the phone
6. Packets and set the VID User Priority and CFI then all the incoming packets will be checked with the IP Address and the VID VID Please set your VID in accordance with your service provider 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 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 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 discarded by the TA The Other incoming packets with different IP address will go through the WAN port to the LAN port VLAN Settings You could set the VLAN settings in this page VLAN Packets Oon Gor VID 802 10 TAG 136 2 4094 User Priority 802 1P 0 l 0 7 CFI 0 0 1 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 en
7. Phone is in use off hook for VolP 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 Or indicates the WAN port is in Connection Flashing indicates the data activity of WAN port 6 Installations 8 SIP Configurations PP GO 10 11 Connect VS211 RJ45 WAN port to ADSL Modem Router using a Category 5 LAN cable Connect VS211 RJ45 LAN port to Notebook PC using a Category 5 LAN cable Connect VS211 RJ11 PHONE port to a Plain Old Telephone Set POTS Connect VS211 RJ11 LINE port to a Public Switched Telephone Network PSTN line Connect the power adaptor 12VDC to power on VS211 and the POWER LED will be ON The PHONE LED indicators will be OFF for about 5 seconds and start flashing for 5 times and remain OFF for VolP configurations The LAN LED will be ON when RJ45 WAN port is connected If the PHONE LED keeps flashing it indicates that VS211 has successfully registered in the SIP server Pick up the phone and the PHONE LED will be Green ON indicating VolP mode If you hear a busy tone please check if the WAN port is connected properly Press 0 keys to PSTN line The PHONE LED will become Yellow ON and you should hear a PSTN dial tone If not please check
8. 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 sign up for a new registered account number Follow the instructions for registration After finished you will receive a mail sent by the INPHONEX mail system and you will get one INPHONE phone number and password in the mail For example the register name phone number is 7123456 with password xxxx Login to the Web configuration page Configurations 4 WAN and LAN Settings WAN Settings You could configure the WAN settings in this page LAN Mode O Bridge NAT IP Type OFixed IP DHCP Client O PPPoE IP Mask Gateway DNS Serverl DNS Server2 MAC Host Name VOIP TA S PPPoE Setting User Name Password 64 LAN Settings You could configure the LAN settings in this page Phone Book LAN Setting Call Settings IP 192 168 123 1 N e Status Mask 255 255 255 0 SIP S WAN Settings MAC 000926ccddee User ee On C Off Save ge Start IP 150 Update End IP po Reboot Lease Time fifo dd hh Submit Reset 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 sip inphonex com After finished the setting click the Submit butto
9. a ring back tone and wait for the called PSTN party to answer Example 9 PSTN to Direct IP Calling Applications The PSTN to Direct IP calling is for the application when the answering parties are with known fixed private or 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 Make a call from PSTN line to the VS211 FXO number e g 7654321 Ina moment you should hear a ring back tone and wait for the VS211 to answer After 4 rings of Auto Answer the FXO port will auto answer with a short beep tone not dial tone Press 12344 for PIN code and then you will hear a dial tone for VoIP mode 4 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 Ina moment you should hear another ring back tone and wait for the VoIP called party to answer 62 Example 10 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 Transfe
10. at Inband DTMF If you are making two stage callings for extension to PSTN you might need to select Outband DTMF option DTMF Setting You could set the DTMF setting in this page Phone Book RFC 2833 Phone Setting O Inband DTMF O Send DTMF SIP Info Submit Reset N2 Codec ID DTMF Rport Settings 8 80 You can enable disable the RPort in this page To change this setting please follow your ITSP suggestions When you finish the setting please click the Submit button RPort Setting You could enable disable the RPort setting in this page RPort SOn OOff Upd Other 38 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 finish 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 O defined by Server 2 Keep Alives Period 60 Default 60 Range 15 250sec that is send keepalives messages in the RTP stream to keep NAT open 3 Jitter Buffer 1 Default 1 Range 0 250 packets SIP Server Type General Asterisk BroadWorks Nortel Xener Vodtel 5 SIP VID 0 2 4094 0 disabled This VLAN ID is for SIP only through WAN RTP VID 0 2 4094 0 disabled Th
11. fixed real IP or private IP under NAT router The SIP to PSTN calling works when both calling and answering parties are registered to SIP server with given registered phone numbers Configurations 1 Same as in Example 2 2 Select ON in the STUN setting page if required by your ITSP 3 Select ON for the Auto Answer and PIN Code in Phone settings Set the Auto Answer Ring Counter e g 3 and the PIN code e g 1234 4 After registration the PHONE LED will be flashing to indicate a successful SIP registration Callings 5 Pick up the phone for VoIP mode and press 16884 or 1688 to call the party with the registered SIP phone number 1688 6 After 4 rings of Auto Answer the FXO port will auto answer with a short beep tone not dial tone Press 1234 for PIN code and then you will hear a dial tone for PSTN Note you must add the postfix 4 PIN Code is used to prevent from call piracy Incorrect PIN Code will result in call disconnect If PIN code is OFF the caller may press PSTN number directly 7 Press 1234567 to call the PSTN party number of 1234567 59 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
12. for HTTP download or 3 Select TFTP and enter the IP address of TFTP server for firmware download then click the Update button Update Firmware You could update the newest firmware Phone Book Method GO Local PC OTFTP Phone Setting Neissik Local PC Code Type CPU xxxx gz gzh SIP Settings File Location aroma NAT Trans Others TFTP TFTP Server 1192 168 1 250 User Password Save Change Upd New Firmware Auto Update W Default Settings Local PC Code Type CPU xxxx gz gzh i File Location on TFTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Upd New Firmware Reb Auto Update A Default Settings Windows Internet Explorer Update Firmware You could update the newest firmware Method OLocalPC TFIP Local PC Code Type CPU xxxx gz gzh File location ee TFTP Server 192 168 1 250 Update Reset 48 Firmware List You could choose one of the firmware to update no Risc Version List vs2ll O2ba gz O vs211 02fa gz oo N mm QU ND Aa O No DSP Version List phone ds O 4 om k ON A O Submit Reset NOTE Do NOT power OFF the VS211 after clicking the Select button or you may damage the VS211 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 t
13. nnne aran nn r narran 37 DIME SENGS tactica E a a a 38 RPO SEWING AAA a aa aia aea aaa S EE 38 Other Setting acta a a e eae i a a E e AEA 39 NATTENS AT OA i ne AAAA AAE A A en 40 STINE 40 Me 40 AUto O 0 y O aa a a a A A a AA 40 EXS amp FXO POM siciassisseisncinsciasdienciansiassiapsdaseiatsiosejatsioasjatsiapaieiadayeieejaaieiaiabeieieiabelsieissansiedentees 43 A ananasen feiende lede fakrinde ina 43 TOME ICAA E EE E E EE EEE EP Pa 44 Adv nCed sspeninsedsndnse densitet lecitina haien 44 STAS a minmin ina reine teaser ansees a a biene diapers 45 User Password miseen ninme a a a a incremental aiii 46 SAVES GHAMGESS OO OPC On aa aia aa a eiet tribe A a A 46 Update EE Aaa a a O A TE A A eae 47 New FintiWaressiisiaiascisseieiciaveiarciereiansiavaiarsiavaiarsiavaisiaranarsieraasieieassagaeauaasseadaanagabadaeasebatanasabaauaaes 47 Auto Update noinine naa A N A A A E E A A 50 Default O 54 NODO poco 54 9 Configurations by Telephone amp lVR oo oocccnnccnnonoocccccnononononnnnnncnncnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnno 55 10 VolP Applications Examples ociosas 57 Example 1 Public Switched Telephone Network PSTN Calling Answering 58 Example 2 SIP to SIP Calling ANnsweriNQ ooooccnnnncnnnncccocccnnnnccnannnonnnccnnn conan conan nn nn cn nan nnn nn nccn nn nncannnns 58 Example 3 SIP to PSTN Calling oooocoonnnccnnnccnnnncconnncnonccnnnnccnnnnnn ease nn cnn naar nn n cnn nn nr anne nn nn nana n cananea 59 Example 4 S
14. 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 Note that the TFTP download works only for public IP address 40 Phone Book Phone Setting Network SIP Settings NAT Trans Othi Auto Config Usel FXO amp FXWPort a MAC Clone Nav Tone Pd Advanced Reb Status Log TFTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Oth Auto Config FXO amp EXWPort MAC Clone Advanced Reb Status Log Auto Configuration Setting You could enable disable the auto configuration setting in this page Auto Configuration of OTP OFP OHTIP DHCP TFTP Option 66 Disable Enable TFTP Server Cu TETP Fie Path Je dommen HTTP Sener IO Exp 60 35 187 30 FTP Sener Po Ep 60 38 17 Auto Configuration Setting You could enable disable the auto configuration setting in this page Auto Configuration Oof TIP OFP OHTIP DHCP TFTP Option 66 Disable Enable TFTP Serer 192 168 62 128 HTTP Sener Loo Exp 60 35 187 30 FIP Saver og ene o Pi 41 FTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Oth Auto Config FXO amp FXWPort _ MAC Clone Advanced Reb Status Log HTTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Oth Auto Con MAC Clone Sav Tone Advanced R
15. 00 respectively Port Settings You could set the port number in this page Phone Book SIP Port 15060 0 65533 Set 0 for auto range as bellow Phone Setting RTP Port 20000 0 65533 Set 0 for auto range as bellow Network SIP Port Range 110000 10999 1024 40000 SIP RTP Port Range 20000 21999 1024 40000 NA En Cod Submit 35 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 Codec Settings You could set the codec settings in this page as Phone Setting Codec Priority 1 Network Codec Pronty 200 6711 atan ARA SIP Service Domain a Farr gt Code WD Codec Prionty 6 6725 24 v lor ME se iy c 7250 NN Codec Prony GSM G 711 amp G 729 G 723 5 3K G 723 5 3K Oon Oof Voice VAD Voice VAD Oon Gof 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 Codec ID Setting You could set the value of Codec ID in this page Codec Type ID Default Value G726 16 ID 23 95 255 M 23 2 95 255 M 2 RFC 2833 ID 101 95 286 101 _ Codec X Service Domain 37 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
16. 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 2 Login VoIP Gateway Username Password pra E Remember last login 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 VolP applications chapter Step 3 Making Point To Point SIP Calls 1 While the PHONE LED is flashing continuously showing a successful registration in the SIP server you may pick up the phone and should hear a dial tone 2 Press 1234564 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 shooting
17. 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 Login VoIP Gateway Username Password T om l Remember last login 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 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 System Information This page illustrate the system related information Model Name Firmware Version Codec Version Protocal Realm 1 Realm 2 Realm 3 MAC address IP address Netmask Gateway DNS IP address Netmask Model Name Firmware Version Codec Version Protocol Realm 1 Realm 2 Realm 3 MAC addr
18. 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 CAN NOT ACCESS WEB PAGE IE Web Browser is a useful tool to configure VS211 When you have difficulties in accessing the default IP address http 192 168 123 1 of VS211 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 VS211 IP address to the same subnet as PC and NAT router ADSL Modem NAT Router WAN LINE VS211 IP 192 168 123 1 j LAN PHONE Example To change VS211 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 VS211 will reboot and LED will start flasing to get an IP address from NAT DHCP server 2 Pick up the phone and press 120 to obtain the VS211 IP address from telephone IVR for example 192 168 62 51 3 Enter from IE web browser http 192 168 62 51 to login VS211 web page for configurations 68 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
19. 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 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 VV VV VV VV VV v v v v v WV 3 Standard Compliances The VS211 VoIP TA supports for the following standards VolP 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 Indicator On the front panel of VS211 there are three LED indicators as the following POWER Or indicates the power is normal PHONE Green On indicates the
20. G MSG 401 is received Use FXO amp FXS Port lt 2005 01 01 00 00 gt REG MSG REGISTER is sent EE lt 2005 01 01 00 00 gt REG MSG 100 is received Say MAC Clone 2005 01 01 00 00 gt REG MSG 200 is received lt 2005 01 01 00 00 gt Reg Status REGISTERED 2005 01 01 08 00 gt Get SNTP server IP 216 184 20 82 lt 2008 11 17 18 25 gt Get Time from SNTP server Succeed Status Log 45 User Password 8 90 You may change the login name and password in this page User Password You could change the login username password in this page New username Phone Book New password Phone Setting Confirmed password Network SIP Settings NAT Trans Others User Password Save Change Save Changes 8 91 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 Save Changes You have to save changes to effect them Phone Book Save Changes Phone Setting Network SIP Settings NAT Trans Others User Password Save Change 46 Update New Firmware 8 92 VS211 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 stored in TFTP server 1 Select the firmware code type Risc or DSP code mostly for Risc code 2 Click the Browse button to choose the updated file location
21. IN If the incoming call is from PSTN the VS211 will answer with a dial tone and allow caller to dial to another VolP number Both Auto Answer function are enabled for Both IP IN and FXO IN Trunk Gateway When the incoming call is from the Internet the FXO port will relay the calling number directly to PSTN without intermediate dial tone This works only for incoming call from VolP PIN Code PIN Code is used to prevent from call piracy If the PIN Code is enabled you will hear double beep tone after calling 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 Call Forward SNTP Auto Answer Caller ID SPP Caller ID Auto Answer You could enable disable the auto answer in this page Auto Answer of OIPIN OFXOIN OBoth O Trunk Gateway Auto Answer Counter 0 8 PIN Code Enabled OF O On PIN Code 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 After you finish the setting please click the Submit button Phone Book Pho Call Forward Caller ID Usej Dial Plan Caller ID Setting You could enable disable the caller ID setting in this page Caller ID Caller ID after 1st Ring FSK Single Caller ID OYes ONo CID Without Time OYes ONo CID Type 2 OYes ONo Dial
22. IP to Direct IP Calling oooonninnnccnnnnicicnnccocccccnnncconnnanrn cnc cnn cnn naar nn ncnn narran nnnn nn nccn nn ncnannnes 60 Example 5 PSTN to SIP Calling ooooocconninonccccnnnnccnnncconccccnnnconnnnn non ee eee eet nn nn cnn nn rca nnnnn nn nn cn nan ccnnnnes 60 Example 6 Direct IP to Direct IP Calling ANnswerinQ oooocccnnnnininnncconcccnnnccnnnnnannn cnn nn cnn nana n ocn nnnnnccnnnnns 61 Example 7 Direct IP to Direct IP Calling within NAT Router rrnrrrnnnnnnnnonnnnrrrrnnnnnnnersnnrrrrnnernnnrnrnnnnr 61 Example 8 Direct IP to PSTN Calling srrrrrnnnnnnonnnnnrrrnnnnnnnrnnnnnrrrrnnnnnnnnnnnnrrrnneeeennennnnnrrnneeeennrnnnnnnn 62 Example 9 PSTN to Direct IP Calling srrrrrnnnnnnnnnnnnrrrnnnnnnnnnnnnnrrrnnnnnnnrrnnnnrrrrneeennnennnnnrrnneesesnnnnnnnnn 62 Example 10 3 Way Conference Call Call Transfer Call Waiting Hold rrrnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnr 63 Example 11 SIP to SIP Calling for http Wwww inphonex COM coooccooccccnnnccconnnannnccnnnnconnnnnn nn ncnnnnccnnnns 64 11 Trouble Shooting for Web Configurati0NS ooccccnnconcccccnnncccnnnnnnannccnnnnnnnnnnnn nro ncnnnncnnnnnnnnnnnnos 68 11 1 DO NOT HEAR DIAL TONE ooococccccccooccncononnnnnonnnncnnnnnnnnnnnnnnn nn rca nan n rra nnnnnnrnnrnnnnnnannnnnanannnnnininns 68 11 2 CAN NOT ACCESS WEB PAGE oooooccocccconocnccconnnncnncnnnncnnconnnnnnnonnnnnnnnnnn oran nnnnnnnnnannn rra nannnnninnnns 68 11 3 ONLY ONE IP AVAILABLE FROM ADSL CABLE SERVICE PRO
23. Law 2 G 711 a Law 7 3 G 723 1 4 G 729a Setting Set Codec 130 1 8 5 G726 16K 6 G 726 24K 7 G 726 32K 8 G 726 40K Seting Set Handset Gain 131 00 15 Ex 13107 and default is 06 Seting Set Handset Volume 132 00 12 Ex 13209 and Default is 10 0 Disable 1 TFTP mode 2 FTP mode Seng at Configure Mode 150 0 3 3 HTTP mode 55 Setting Enable Call MAR 138 This will disable Call Transfer Setting Disable Call Waiting m3 fhs will enable Call Transfer You must unlock keypad first in order to Setting Unlock Keypad 190 change settings by keypad Setting Lock Keypad mow Keypad can NOT be used for setting After you hear Option Successful hang up TA will reset back to factory defaults Setting Factory Reset 198 WARNING ALL User Changeable NONDEFAULT SETTINGS WILL BE LOST Setting Blind Call Transfer 510 XxXXXXX Ex 510 54321 transfer to 54321 Setting Attendant Call Transfer Aston Ex 115114543214 transfer to 54321 Setting 3 Way Conference Call Ex 45124543214 conference with 54321 Attendant Call Transfer to Ex 514 12345678 transfer to 12345678 Setting PSTN FS OO ODOT from IP to PSTN Line Switch to PSTN port will hear dial tone of PSTN Line 56 10 VolP Applications 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 vie ADSL Modem o
24. Make sure the PHONE LED is flashing continuously with a successful SIP registration Callings 3 Pick up the phone for VolP mode 4 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 Example 5 PSTN to SIP Calling Applications The applications can be for ADSL connections as in both Diagrams A and B Both parties are registered to SIP server with either fixed real IP or private IP under NAT router Configurations 1 Same as in Example 2 2 Select ON for the Auto Answer and PIN Code in Phone settings Set the Auto Answer Ring Counter e g 3 and the PIN code e g 1234 3 Make sure the PHONE LED is flashing continuously with a successful SIP registration Callings 4 Make a Call from PSTN line to the VS211 FXO number e g 7654321 In a moment you should hear a ring back tone and wait for the VS211 to answer After 4 rings the VoIP mode will auto answer with a short beep tone not dial tone Press 1234 for PIN code and then you will hear a dial tone for VolP mode Incorrect PIN Code will result in call disconnect Note that if PIN code is OFF there will be not short beep tone and the caller may press SIP number directly 5 Press 16884 or 1688 to call the party with the registered SIP phone number 1688 In a moment you should hear a ring back tone and wait for the VoIP c
25. P 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 Please refer to Example 11 for more detailed SIP server registrations Configurations 1 Select either NAT or Bridge in accord with your network in WAN settings page N Select DHCP Client to automatically get an IP address from NAT router oo Remember to click the Submit button Select Active ON in the SIP settings Service Domain page 58 5 Enter the items of Register Name Register Password Proxy Server and Outbound Proxy 6 Select ON in the STUN setting if required by your ITSP 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 VolP mode 9 Press 1688 or 1688 to call the party with the registered SIP phone number 1688 Note 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 Both parties are registered to SIP server with either
26. 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 Routing to For selection of FXO the call will be routed from FXS port to FXO port PSTN when the dialing number matches the Routing rule If not matched the call will be routed to VolP For selection of IP the call will be routed from FXS port to VolP if matched and vice versa Routing rule This defines the routing rule from Telephone FXS to either PSTN line or VoIP This is convenient for dial plan without using 0 function key 8 26 Symbol Representations Symbol Representations 0 1 2 3 4 5 6 7 8 9 or D It Drop the matched dialing prefix number This is for routing rule only Example 1 Route to FXO Routing rule D02 0800 a Pressing 02xxxxxxxx will result in dialing out to PSTN port 2200000XX b Pressing 0800xxxxxx will result in dialing out to PSTN port O800xxxxxx c Other dialing number will route to VolP and follow the rules of Drop Prefix Example 2 Drop Prefix
27. VIDER cooooooccccccconccnonannccncnnn 69 11 4 VOIP EXTENSION CALLS TO PSTN ARE NOT WORKING c oooooococcccococccccococnconconnnnononccnnnncons 70 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 VolP 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 VolP 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 VolP TA is equipped with two RJ11 connectors and two RJ45 connectors and is featuring as the following gt Three LED Indicators for VS211 POWER PHONE LAN RJ45 x 2 for WAN and LAN ports RJ11 x 2 for
28. VS211 1 FXS 1 FXO SIP VolP Telephone Adaptor User Manual V2 1h Quick Guide Step 1 Broadband ADSL Cable Modem Connections for VS211 Connect VS211 WAN port to ADSL NAT Router as the following connection Connect VS211 LAN port to Notebook PC LAN port using a Category 5 LAN cable Connect VS211 RJ11 PHONE port to a Telephone Set Connect VS211 RJ11 LINE port to a PSTN Telephone Line Connect 12VDC 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 F The PHONE LED will be Green flashing for a successful SIP registration Pick up the phone and it will be Green ON indicating VolP 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 LAN IP address e g 192 168 123 1 for VS211 After the IP announcement please hang up mUo Figure A ADSL Connections with NAT Router for VS211 ADSL Modem NAT Router VS211 IP 192 168 123 1 WAN p LINE PC IP 192 168 123 150 O T PHONE A a Step 2 Settings for VS211 from PC Web Browser A VS211 is defaulted at embedded NAT mode B Press 120 from the phone to listen to IVR and take down 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
29. abled the VLAN will only function on ATA The packets from LAN port will pass through WAN without VLAN function 28 DMZ 8 56 The DMZ can be enabled disabled and configured in this page DMZ Setting You could configure your demilitarized zone setting in this page Phone Book DMZ Oon Gor Phone Setting DMZ Host IP 0 0 0 0 DDNS Oth TLAN Usei DMZ Sav Virtu Server 29 Virtual Server 8 57 The Virtual Server IP and Port numbers can be configured in this page Virtual Server Settings You could set your virtual servers in this page The usual port numbers are WEB TCP 80 FTP Control TCP 21 FTP Data TCP 20 E mail POP3 TCP 110 E mail SMTP TCP 25 DNS UDP 53 and Telent TCP 23 Phone Book Virtual S r Page Phone Setting 0 O EA NAO A 2 EE E HIT A Fr AAA a 4 JO O BT TA A AO AO A 6 3 O Add Virtual Server Sener IP Protocol TCP MM Internal Port Start IntemalPortEnd External Port Stat ExtemalPortEnd 30 L2TP 8 58 The Layer 2 Tunneling Protocol L2TP Server can be set ON OFF in this page This L2TP can be used to traverse the firewall when used with Virtual Private Network VPN applications L2TP Settings You could set the L2TP server in this page Phone Book L2TP Oon Gor Phone Setting Net Status 31 PPTP 8 59 The Point to Point Tunnel Protocol PPTP Server can be set ON OFF in this page This PPTP can
30. alled party to answer 60 Example 6 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 VolP 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 VolP called party to answer Example 7 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 Enter the items of IP Subnet
31. be used to penetrate the firewall when used with Virtual Private Network VPN applications PPTP Settings You could set the PPTP server in this page PPTP Oon Gor 32 SIP Settings 8 60 You can setup the Service Domain Port Settings Codec Settings RTP Setting RPort Setting and Other Settings for SIP Proxy Server registrations in this page Service Domain 8 61 8 62 8 63 8 64 8 65 8 66 8 67 8 68 8 69 8 70 8 71 8 72 8 73 8 74 8 75 You may register up to three SIP Servers for three Realms in the VS211 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 To select SIP Server 1 default please pickup the phone press 1 then hangup To select SIP Server 2 or 3 please pickup the phone press 2 or 3 then hangup Click Active ON to enable the Service Domain then enter the following items Display Name enter the name you want to display User Name enter the User Name given by your ITSP Register Name enter the Register Name given by your ITSP Register Password enter the Register Password given by your ITSP Domain Server enter the Domain Server given by your ITSP Proxy Server enter the Proxy Server given by your ITSP Outbound Proxy enter the Outbound Proxy of ITSP If not provided you may skip this Register Per
32. conds default wait time the VS211 will automatically call the preset IP or phone number 22 Alarm 8 37 You can configure the Alarm setting in this page Alarm Settings You could set the alarm time in this page Phone Book Phot Call Forward SNTP Alarm Time o Ho hh mm Alarm OON OFF Current time 2008 11 17 15 12 Flash Time D Call Waiting T 38 FAX Alarm Y Network 8 38 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 39 You can check and show the current Network settings in this page Interface 0 is for WAN port Status and Interface 1 for LAN port Status 23 Network Status This page shows current status of network interfaces of the system Phone Book System Up Time Phone Setting Network Link Up Time NAT Type Type DNS Server 1 DNS Server 2 Type IP Mask Gateway DNS Server 1 DNS Server 2 WAN 0 day s 0 hour s 4 minute s 0 day s 0 hour s 4 minute s Port restriced cone Interface 0 DHCP Client 192 168 62 111 255 255 255 0 192 168 62 1 168 95 192 1 168 95 1 1 Interface 1 DHCP Serve
33. eb Status Log Auto Configuration Setting You could enable disable the auto configuration setting in this page Auto Configuration Oof OTP G FP OHTIP DHCP TFTP Option 66 Disable Enable niese 0 TFTP FilePath xp downoad HTTP Sener po Exp 60 35 187 30 1921681250 See FTP Password A Auto Configuration Setting You could enable disable the auto configuration setting in this page Auto Configuration Oot OTP OFP HTIP DHCP TFTP Option 66 Disable Enable TFTP Sener Ld TFTP Fie Path Eos voip sipsener com Exp 60 35 187 30 a BE FTP Sener Lo pan 42 FXS amp FXO Port 8 85 You may select the FXS and FXO impedence of the analog telephone by different countries FXO FXS Setting You could select the FXO amp FXS impedence of the analog telephone by different country in this page Phone Book Phone Setting FXO Port USA v MED usa w FXO Silence Timeout 30 1 250 minutes FXO CID forward Oon Oof FXO amp FXS Port MAC Clone o 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 finish settings please click the Submit button MAC Clone Setting You could enable disable the MAC clone setting in this page Phone Book MAC Clone Oon Gor Phone Setting Network SIP Settings NAT Trans FXO amp FXS Port MAC Clone Tone 43
34. ending anything Disable disable IP dialing function 5 Encryption Type Must be disabled for standard SIP protocol Please consult your ITSP for encryption type when registered in ITSP 6 Encryption Key Special key for encryption need 44 Advanced Setting You could change advanced setting in this page Phone Book Phone Setting Send Anonymous CID OYes ONo Stop feature tone OYes No MMI forward block Network SIP Settings Bling Sgn NAT Trans CPC Delay e 2 5 Seconds Oth Auto Config Use FXO amp FXS Port Cee Duration o x 10 ms 0 12 IP Dialing format Type 1 x x x x x Y Sera isher o Encryption Type PPPOE rety perio En System Log Type none AA _ MAC Clone Status Log 8 89 You can check system status log Status Log lt 2005 01 01 00 00 gt Application starting lt 2005 01 01 00 00 gt Init Wan Interface Phone Book lt 2005 01 01 00 00 gt Iface type DHCP_CLIENT lt 2005 01 01 00 00 gt Init Lan Interface Phone Setting lt 2005 01 01 00 00 gt Enable DHCP_SERVER lt 2005 01 01 00 00 gt Iface type FIXED IP Network lt 2005 01 01 00 00 gt DHCP_SendDiscover i lt 2005 01 01 00 00 gt Rx OFFER from 192 168 62 1 SIP Settings lt 2005 01 01 00 00 gt DHCP_SendRequest E lt 2005 01 01 00 00 gt Got DHCP Ip 192 168 62 110 NAT Trans lt 2005 01 01 00 00 gt REG MSG REGISTER is sent Oth 2005 01 01 00 00 gt REG MSG 100 is received 2005 01 01 00 00 gt RE
35. ess IP address Netmask Gateway DNS IP address Netmask VolP FXS FXO TA Fri Apr 24 13 18 29 2009 02hb Mon Apr 20 10 27 21 2009 Call Status SIP Not Registered Not Registered Not Registered WAN Status 0009261002d4 192 168 62 111 255 255 255 0 DHCP Client 192 168 62 1 168 95 192 1 168 95 1 1 LAN Status 192 168 123 1 255 255 255 0 DHCP Server Show device Model Name Device Risc version eg Tue Jan 16 11 28 32 2007 Device DSP version eg Wed Dec 20 17 28 06 2006 Call Status Show device VoIP protocol Show the first registry information status Show the second registry information status Show the third registry information status WAN port of MAC address IP address netmask Default router IP address DNS IP address LAN Status Current LAN port IP address netmask 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 se
36. he available firmware for download 8 94 Select the new file you want to download to the VS211 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 49 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 Auto Update Settings You could set auto update settings in this page Phone Book Update via of OTP O FTP OHTIP Phone Setting Memel TFTP Server SIP Settings TFTP File Path Exp download NAT Trans HTTP Server Exp 60 35 187 30 Others r HTTP File Path Exp download User Password Save Change FTP Server Exp 60 35 17 1 Upil New Firmware FTP Usemame Auto Update FTP Password Default Setilllgs FTP File Path Exp file load Check new firmware Power ON and Scheduling Scheduling only Scheduling Date 14 1 30 days Scheduling Time AM 00 00 05 59 Y Automatic Update Notify only O Automatic Firmware File Prefix TA1S10 Next update time Notes Check new Firmware Power on and Scheduling Check new firmware whe
37. if the PSTN Line is connected Press 111204 to check the assigned IP address for the VS211 The default IP address is 192 168 123 1 You may enter this IP address in IE Web browser from Notebook PC Please refer to Chapter 8 for web configurations Register VS211 into your SIP server Please refer to VolP applications examples of SIP registrations After successful registration to the SIP server the PHONE LED will start Green flashing Pick up the phone and you should hear a dial tone Press 123456 to call the party with the number 123456 registered in the SIP server In a moment 5 seconds you should hear a ring back tone and wait for answer Note that you may press 1234564 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 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 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 VS211 has successfully registered in the SIP server 8 Configurations by Web Browser Login VolP Gateway You may enter the IP address from PC Web browser to configure VS211 For example enter http 192 168 123
38. iod enter the Register Period in minute given by your ITSP When it shows Registered in the Register Status it indicates a successful registration to the ITSP and the PHONE LED will start flashing The VS211 is then ready for VoIP call If you have more than one SIP account please follow the steps to register to other ITSPs After you finish the setting please click the Submit button 33 Phone Setting SIP Service Domain NA Port Codec ID Phone Book Phone Setting Network SIP Service Domain NA Port W Codec ID Usel DTMF Reboot Service Domain Settings You could set information of service domains in this page Realm No Realm 1 ive Oon Gof Registr Pasemor Outbound Proxy Subscribe for MWI Oon Gof Status Not Registered Service Domain Settings You could set information of service domains in this page Realm No Active Oon Gor Display Names User Name E gg RegisterName Register Password E Domain Serer Proxy serer Outbound Pry A Subscribe for MWI Oon Gor Status Not Registered 34 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 finish the setting please click the Submit button The defaults for SIP port and RTP port are 5060 and 200
39. is VLAN ID is for RTP voice packets Other Settings You could set other settings in this page Phone Book Hold by RFC Oon Gor Phone Setting Voice QoS DiffServ 40 0 63 Network SIP QoS Dif Serv 140 0 63 SIP Service Domain SIP Expire Time 60 15 86400 sec O define by Server s Use DNS SRV Oon Dor Send Keep Alives Packet Oon OOF Codec ID Keep Alives Period 60 15 250 sec Use DTMF Jitter Buffer 4 0 250 packets SIP Server type General SIP VID VLAN 0 2 4094 0 disabled Reboot RTP VID VLAN 0 2 4094 0 disabled 39 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 STUN Setting You could set the IP of STUN server in this page Phone Book STUN Oon Gor Phone Setting STUN Server Network STUN Port 80 65535 SIP Settings Force Public IP Oon Oof Public IP address Others IG Port 80 65535 User Password Others 8 83 You can setup Auto Config FXO8 FXS Port MAC Clone Tone and Some Advanced Settings and Status Log 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
40. n 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 65 Service Domain Settings You could set information of service domains in this page Realm No Realm 1 v Active S On O0f spe ra User Name 7123456 Register Name Register Password TT Domansener so prono O Proxy Server sip inphonex com C Subscribe for MWI Oon Gor Status Registered INPHONEX SIP Server Register Name 7123456 Password xxxx Domain Server sip inphonex com Proxy Server sip inphonex com 66 Codec Settings You could set the codec settings in this page Codec Priority 1 A 21 ow gt NNN Codec Priority 3 A c7 Codec Priority 5 CN c 72 2 A Codec Priority 7 GE Codec Priority 9 GSM G 711 amp 6 729 om ooo G 723 5 3K G 723 5 3K O0n off Voice VAD Voice VAD Oon Sor 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 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
41. n power on and in accord with schedule Must be manually updated Scheduling Only default Check in accord with schedule Scheduling Date Default at 14 days Scheduling Time Default at AM 00 00 05 59 randomly 50 TFTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Upd New Firmware Auto Update Y Default Settings Auto Update Settings You could set auto update settings in this page Update via Oof GTFTP O FTP OHTIP TFTP Server 192 168 1 250 mere em epost HTTP Sener ESSE FTP Server Exp 60 35 17 1 Check new firmware Power ON and Scheduling Scheduling only Scheduling Time AM 00 00 05 59 vw Firmware File Prefix TA1S10 Next update time 51 FTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Auto Update Default Settles Auto Update Settings You could set auto update settings in this page Update via Oof OTIP FIP OHTIP TFTP Server TFTP Fie Path p downto HTTP Server Exp 60 35 187 30 192 168 1 250 FTP Password E ne Check new firmware Power ON and Scheduling Scheduling only Scheduling Time AM 00 00 05 59 vw Firmware File Prefix TA1S10 Next update time 52 HTTP Mode Phone Book Phone Setting Network SIP Settings NAT Trans Others User Password Save Change Auto Update Default settles
42. nt 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 1034 on telephone and the TA will call the registered number 612345 Phone Book You could add delete items in current phone book Pho Phone Book Phone Book Page page 1 Y Phone Setting Phone Name Number or URL Network 0 SIP Settings 1 101 192 168 1 100 O NAT Trans 2 102 james sipserver com O 3 103 612345 O Others 4 User Password 5 Save Change 6 Update 7 8 Reboot 9 Delete Selected Delete All 11 Phone Settings 8 14 The subpages are as follows Call Forward SNTP Volume DND Auto Answer Caller ID Dial Plan Flash Time or hook switch Call Waiting T 38 FAX over IP Hot Line and Alarm settings Call Forward 8 15 You can select the forward mode and enter the forward URL All Forward All incoming call will forward to the URL or PSTN 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 URLor PSTN when no answer 8 16 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 Phone B
43. numbers with 9 or more digits are prohibited Auto Dial Timer The inter digit timer Default is 5 seconds None SIP Server Mode When SIP Settings gt 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 When you finish the setting please click the Submit button Click the Save button The changes you have made will be saved and the VS211 will reboot automatically Dial Plan You could the set the dial plan in this page Routingto OIP FXO Disable Phone Book Phot Call Forward Drop prefix OYes No oe Mo Drop prefix Yes ONo Replace rule 2 006 002 003 004 005 007 009 Drop prefix GO Yes ONo Flash Til Replace me 3 009 Call Waiting Drop prefix OYes No Upd 3S FAX j Replace rule 4 Bort 35xx 2 10 Reb Hotline Alarm Dial now XX XX 11x 1 237 XX XXXXXXXX Exp 1 137 XX 345XX 45XX67 Use as send key O Yes ONo wee f 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
44. o the other side s handset PSTN In Gain is to set the volume send out to the other side s handset Volume Setting You could set the volume of your phone in this page Handset Volume 110 0 12 Call Forward PSTN Out Volume 10 0 12 SNTP enk Handset Gain 110 0 15 DND l al PSTN In Gain 10 0 15 Caller ID DND 8 19 You can configure the DND Do Not Disturb setting to keep the phone silence You can choose either DND Always or a DND period 8 20 DND Always All incoming call will be blocked until this feature is disabled 8 21 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 22 After you finished the setting please click the Submit button DND Setting You could set the do not disturb period of your phone in this page Phone Book Pho Call Forward DND Always Oon Of SNTP DND Period Oon Gor From hh mm To oo Hoo hh mm Auto Answer 8 23 Auto Answer function can be used to relay calls between VolP and PSTN When the ring count exceeds the number set in Auto Answer Counter the Auto Answer function will be activated when one of the 4 Auto Answer modes is chosen IP IN When the incoming call is from the Internet the FXO port will answer with a PSTN dial tone and wait the caller to dial another PSTN phone number FXO
45. ook Call Call Forward SNTP Settings sr Veme Soner fae User Save Dial Plan Settings Upda Flash Time Settings T 38 FAX Settings Forward Setting You could set the forward number of your phone in this page All Forward Oof OGIP OPSTN Busy Forward Oof OP No Answer Forward of OP OPSTN EE ATA All Fwd No 161234 61234 Busy Fwd No Jane 192 168 62 100 No Answer Fwd No James james sipserver com No Answer Fwd Time Out 3 2 8 Ring SNTP 8 17 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 SNTP Settings You could set the SNTP servers and Daylight Saving Time DST in this page SNTP S On OOff Pho Call Forward Primary Server north america pool ntp org MES DI x olu Caller ID sm o o Daylight Saving Oon Gor DST ofset Alarm DST Start Date DayofMonth 0 Week of Month IE oo gt DST End Date ee Week of Mont etme SN Time Zone GMT v 08 00 hh mm 38 Volume 8 18 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 t
46. r 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 Attendant Transfer to PSTN Party A calls Party B While in conversation Party B may press Flash key to hold the call and then press 514 PSTN number of Party C to call and talk to Party C Hang up from Party B and Party A will be transferred from FXS port to FXO port and connect to PSTN number of Party C 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 push the Hold Flash key to hold the current call for a while then push Hold key again to resume talking 63 Example 11 SIP to SIP Calling for http www inphonex com Applications
47. r 192 168 123 1 255 255 255 0 192 168 123 1 168 95 192 1 168 95 1 1 8 40 The WAN setting is used to configure the WAN port connects to the ADSL Modem Router 8 41 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 settings will be ignored When setting to Bridge Mode only the WAN settings will get effective and the LAN 8 42 There are three selections for WAN IP Type Fixed IP DHCP Client and PPPoE modes This WAN setting is for the WAN port when set in NAT mode The default is at DHCP Client 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 finish the settings please click the Submit button Phone Book Phone Setting LAN WAN Settings You could configure the WAN settings in this page LAN Mode O Bridge NAT aa IP DHCP Client PPPoE Mask 255 255 255 0 DNS Type Fixed narn DNS Server2 168 95 1 1 MAC 000926100244 Host Name VOIP_TA1S10 PPPoE Setting User Name S Service Name Submit 8 46 The default IP address is 192 168 123 1 with Net Mask 255 255 255 0 and DHCP Server enabled The range of IP addresses for DHCP is f
48. rmer LINE LAN nal PHONE on B ADSL Connections with NAT Router for VS211 A ADSL Modem arenas NAT Router WAN LINE VS211 IP 192 168 123 1 LAN PHONE 57 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 Phone 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 Pick up the phone and press 0 key for PSTN mode and you should hear a dial tone 5 Press e g 1234567 to call the PSTN party with 1234567 In a moment you should hear a ring back tone and wait for the called PSTN party to answer 6 When receiving PSTN incoming call you must pick up the phone within 4 rings to answer otherwise the VolP mode will answer automatically for IP extension call Please refer to Example 5 for more details of PSTN to SIP extension calls 7 If the Auto Answer is OFF the FXO port will become PSTN only and the function of extension call from PSTN to SIP call will be disabled Example 2 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 I
49. rom 150 to 200 8 47 Connect your PC to the LAN port set your PC as DHCP mode and the PC will automatically get an IP address from the VS211 8 48 When you finish the settings please click the Submit button 25 LAN Settings You could configure the LAN settings in this page Phone Book Phone Setting 192 168 123 1 Net Status IE gt IE AN MAC 0009261002d4 DHCP Server Lease Time 1 fo dd hh DDNS 8 49 You need to have a DDNS account before configuring the DDNS setting Usually most of the VolP 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 finish the setting please click the Submit button 26 Phone Book Phone Setting Nety Status Example DDNS Settings You could set the configuration of DDNS in this page DDNS Oon Gor E mail Address a A DONS Sener List Wild Card Off Line On Off DDNS Settings You could set the configuration of DDNS in this page DDNS On Oof Esl ie rea User Name E mail Address DDNS Server DDNS Server List test ce MA members dyndns org Y Ie oors v Wild Card w Of Line Oon Gor 27 VLAN 8 50 8 51 8 52 8 53 8 54 8 55 Phone Book The VLAN setting is for VoIP packets related to LAN port VLAN Packets If you enable VLAN
50. s 99 NIO sd kg TABLE OF CONTENTS MIT ae 6 TT EE ET E E EN Er 6 Standard ComplancaS ninia is 7 PACKING COMIGING asp sees annus reuniuadiaaesansneresanetaneand 7 LED INAICATOT Lende ii 7 Installations amp SIP Configurations ocooooconccccccnnccconnnnanancnonnnccnnnnnnnnnncnnnnnnnnnnnnn nro ncnnnnnnnnnnnnrnnnno 8 Default Reset by Telephone rnnnnnnnnnnnnnnnnnnnnnnnnnnnnnannnnnnnnnnnnnnrnnannnnnnnnnnnennrnnnnnrnnnnnnesensrnnnnnn 8 Configurations by Web B SE sica ad 9 Login VOIP E 1 EN AAA OU de Naa aA 9 SEC 0 sieis aariaa a anaana aaa a Ea aa a aa ana an a a a EE Na Ea NE PARNA TEN A es 10 Phone Book Seting Siinmaa aaar aan Eaa aar e E aaraa T ar E Pa aaa ran Naa iia 11 PHONE BOOK E E 11 Phone SENOS iinom a a a Aa aa aa ETE aE Ea aaa ocre 12 Gall Forward e E onic saa eeesess texte cca teeeees steerer eectams steer oc eeeseamee 12 NR NNN 13 vme eck ctccceccctcacecectcactccctaactuacteactaacteactenctcacanacteccicaateseicicteieeaicteietatcees 14 ND 15 AUTO ANTENT 15 clerDa cra AAA 16 Dialnet tete adenda bi 17 Femme 19 cal Mama 20 CallTranstertincione craneal T E E 21 TRINN 21 HOM tit n 22 FET AAA EEEE A AEA 23 NOWO onenn EEEE NRE 23 NEtWOIK SAUS ae AE 23 e 24 A a 25 e 26 A a 28 BIN VA EE OE 29 ERA AEG 30 er EEE 31 PPT Reneon nanan iR EESE EENE Gaaadedetasstadetaaadadataaadadatnaete 32 A 33 Service DOM A rr nakne nb ke id are d d ie 33 A A ee ee 35 E A A at aat 36 Codec ID Settings ron r rn cnn rn nn rn
51. the call waiting function When you finish the setting please click the Submit button Flash Time Setting You could set the flash time in this page peta Pho Call Forward Generate Flash Signal 10 x 10 ms 9 120 ND S 3 Plash Sanai Dei WAN 60 tome 4 258 Auto Answer Flash Signal Detect MIN 7 x 10 ms 3 12 Dial Pian Flash Time Call Wait Notes 1 Flash Signal Detect MAX Maximum Flash Hook duration unit 10ms 2 Flash Signal Detect MIN Minimum Flash Hook duration unit 10ms 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 finish the setting please click the Submit button Call Waiting Setting You could enable disable the call waiting setting in this page Phone Book Call Waiting SOn Oof AY Auto Answer Fast Time Call Waiting T 38 FAX W 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 b
52. 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 LAN PC IP 192 168 123 150 Example To change PC IP address to the same subnet as 192 168 123 1 for VS211 1 Oo ON DO oO Q 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 192 168 123 150 150 can be any number other than 1 which is used by VS211 Enter Subnet mask 255 255 255 0 Enter Default gateway 192 168 123 1 Click on OK button You will lose internet connection at this time At IE browser type http 192 168 123 1 Follow the example in Advanced Settings for Embedded NAT for web login At LAN setting turn on DHCP server AtWAN setting choose DHCP client to work with your ADSL Cable modem Save change wait for VS211 to reboot Change your PC s Internet Protocol TCP IP Properties back to obtain an IP address automatically 10 You may press 120 and 126 to listen to the IVR for LAN and WAN IP addresses 69 11 4 VOIP EXTENSION CALLS TO PSTN ARE NOT WORKING You must enable the Auto Answer function in Call Se
53. ttings in order to answer automatically by VolP 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 VolP extension call Extension Calls from Internet VolP to PSTN If the incoming call is from VolP then FXO port will answer with a PSTN dial tone and allow caller to redial to PSTN phone number Extension Calls from PSTN to Internet VolP If the incoming call is from PSTN then VS211 FXO port will answer with a short beep tone and allow caller to redial to VolP Phone number 70
54. y enabling the call waiting function 41388 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 u Law or a Law and make sure your SIP server gateway also supports this T 38 function When you finish the setting please click the Submit button T 38 FAX Setting You could enable disable the FAX function in this page Phone Book T 38 FAX Con Oof Phoi Call Forward T 38 Pass throug codec uLaw OaLaw SNTP Caller ID Usej Dial Plan Flash Time Sav Call Waiting T 38 FAX Hot line w 21 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 VS211 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 Phone Book Pho Call Forward om __ FashTine Hot line Alarm Ww Hot line Setting You could set the hot line in this page Use hot line O Enable O Disable Hot line Number Pick up the phone In 1 2 se

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