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Fanvil Product User Manual IP Phone Model: BW210
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1. This VoIP Phone Supports DHCP by default It will receive an IP address and other network related settings Netmask IP gateway DNS server from the DHCP server If your network supports DHCP you can connect this VoIP Phone directly to the network If your network doesn t support DHCP you need change this VolP Phone s network connection setting According to the following steps change this VolP Phone s DHCP network connection setting into PPPoE or static IP which your network supports at present 2 2 1 PPPoE mode 1 Press the 3 key for three seconds then confirm it by the Enter key your phone network connection mode will switch into PPPoE mode Prepare your PPPoE account name and password 2 Press the MENU key the LCD screen will display INPUT PASSWORD 3 Input the password default value is 123 and press the ENTER key the LCD screen will display NETWORK Enter 4 Press the key and LCD screen will display LAN press the key enter it by Enter the key the LCD screen will display STATIC NET Then press the key Enter again enter it by the key the LCD screen will display USER NAME Enter 5 Press the e and then press the a key input your PPPOE account number then press the key to confirm The LCD screen will display the inputted PPPOE account number Exit aD 6 Press the key to return to the previous menu then press the key the LCD screen will di
2. Press the button then the gt button you will enter into the phone book R Send Press the E 7 button to select your desired contact person then press the button to dial the call eo Use Callers Press the key then select your desired phone number in callers by the R Send E 7 key and next press the Use the R send key button to dial the call Please pick up or press the E key After you hear dialing tone please press the key to dial the last phone number Note after you reboot the phone the phone will delete callers and Redial will be invalid 3 1 3 Ending a call e Hangs up by handset onhook e Hangs up by press O when in hands free e Hangs up a call in call waiting state If you are in call waiting state you could press key to hang up the current call and switch to the other call to keep talking Note Pressing key will not hang up if there is only one call currently 3 1 4 Transferring a call Call transfer has several ways to realize 1 When A talks to B B may press the key and dial C phone number After B talks to C or B hear alert from C B presses the UGG key then B hangs up and A will get through to C 2 When A is talking with B C calls B B may press the key to hold A and talk to C Then B presses the key A will get through to C 3 When A talks to B B presses the WAB amp key dial C phone number and key then hang up and A will get through to C 1 a
3. 0 0 0 0 if you press key to display the current IP if the icon DHCP is showed without blink it means that the phone has already gotten IP from DHCP server 3 Basic Functions 3 1 Basic operation 3 1 1 Accepting a call There are four methods to accept an incoming call Pick up handset to accept incoming calls o o Press the button e If you need switch from a hands free call to handset please pick up the handset directly o e If you need switch from a handset call to hands free please press the and then hang up the handset button 3 1 2 Making a call O Quick dialing 11 In idle mode input the called number and press key or button phone will dial the call and use hands free automatically e Use handset Pick up the handset and the LCD screen will display PLEASE DIAL and you will hear dialing tone at the same time then input the phone number and end by the button When you hear long ring du du from handset and the LCD screen display CALLING the call is through Hang up the handset to end the call eo Use hands free o Press the button and the LCD screen will display PLEASE DIAL and you will hear dialing tone at the same time then input the phone number and end by the button When you hear long ring du du and the LCD screen display CALLING the call is through Press the Use the phone book button again to end the call Enter
4. 1 Special Keys e Realize Secondary Dial by Dialing for only one time When you make secondary dial in off hook handsfree standby pre input mode press key to postpone input and screen display will show One stands for 2 seconds For example you input 123 45 the phone will send DTMF 45 2 seconds after the phone call 123 123 45 will make phone send DTMF 45 at 6 seconds interval o MWI Message Waiting Indication When a new voicemail coming LED on the phone will flash You can press the MWI key to listen new voicemail if you configure mwi number 3 2 2 Call pickup Call pickup is implemented by simulating pickup function of PBX it s that when A calls B B 14 rings but no answer at this moment C can hook off and input an appointed prefix plus B s number pick up A s call and talk with A The following chart shows how to configure an appointed prefix in dial peer to have call pick up function 1 means appointed prefix code After making the above configuration C can dial 1 plus B phone number to pick up A s call User can set prefix in random in the case of no affecting current dialing rules 3 2 3 join call When B is calling C A can join in the existing call by inputing an appointed prefix numbers plus B or C number if B or C also supports join call The following chart shows how to configure an appointed prefix in dialpeer to have join call function rep joincall nosuffix B 2 means
5. 168 1 118 for the out_access rule But if devices ping other devices which network ID is 192 168 1 0 it will be normal Rule Delete Index To Be Deleted Click the Delete button to delete the selected rule 4 3 7 Logout System Logout logout Press the Logout button to Logout Phone Click Logout and you will exit web page If you want to enter it next time you need input user name and password again 43 4 4 Settings via phone s keyboard 4 4 1 How to set via the phone s keyboard Press Menu Up Down Enter and exit key to browse select and cancel Use the Up Down key to browse the menu and submenu Use the ENTER key to enter into submenu and confirm your operation the EXIT key can be used to back and cancel operation 4 4 2 Phone menu Phone main menu 44 5 Appendix 5 1 Specification 5 1 1 Device specification item this VoIP Phone Adapter Input Output 1 out 100 240VAC 50 60Hz Output 5V 1A prg m 10 100Base T RJ 45 for LAN Auto MDIX 10 100Base T RJ 45 for PC Auto MDIX Power Consumption Idle 1 5W Active 1 8W 74 x 28mm Operation A Relative Humidity 10 65 Main Chipset SDRAM 8Mbits gt gt gt Size W x H x D 11 6x8x3 in 295x205x75mm 2 071b 0 94kg 5 1 2 Voice Features e Support 2 lines SIP SIP 2 0 RFC3261 e Codec G711A u G 7231 high low G 729 G 722 G 726 e Echo cancellation Support G 168 and hand free can
6. Firewall is also called access list is a simple implementation of a Cisco like access list firewall It supports two access lists one for filtering input packets and the other for filtering output packets Each kind of list could be added 10 items We will give you an instance for your reference 42 Deny Permit oeny v ear PortRonge moretones esk Field name explanation Select it to Enable in_ access rule Select it to Enable out_ access rule output rule Permit rule Protocol Type Filter protocol type You can select TCP UDP ICMP or IP Port Range Set the filter Port range Md Set source address lt can be single IP address network address complete address 0 0 0 0 or network address similar to 0 Set the destination address It can be IP address network address complete address 0 0 0 0 or network address similar to Set the source address mask For example 255 255 255 255 means just point to one host 255 255 255 0 means point to a network which network ID is C type Set the destination address mask For example 255 255 255 255 means just point to one host 255 255 255 0 means point to a network which network ID is C type Click the Add button if you want to add a new output rule Firewall Output Rule Table Then enable out_access and click the Apply button So when devices execute to ping 192 168 1 118 system will deny the request to send icmp request to 192
7. Static Mode setting please ask your ISP for them e DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artificially PPPoE In this mode your must input your ADSL account and password You can also refer to 3 2 1 Network setting to speed setting your network Choose Static IP MODE click NEXT can config the network and SIP default SIP1 easily also can browse them too Click BACK can return to the last page Static IP Set A sss Prmarvons RR S S TT sms O Netmask ImputtheNetmask distributed to you DNS Domain Set DNS domain postfix When the domain which you inputted can not be parsed phone will automatically add this domain to the end of the domain which you inputted before and parse it again AlterDNS Input your standby DNS server address hone Number username po S Enable Register Display Name If user set the display name callee will show this display name UserName T Inputyour SIP register account name Phone Number Input the phone number assigned by your VOIP service provider Enable Register Start to register or not by selecting it or not Connect Mode Static Static IP Address 192 168 1 179 Register Server 192 168 1 2 HO Display detailed information that you manual config Choose DHCP MODE click NEXT to config simple SIP default SIP1 You can browse it too Click BACK to return to t
8. it the phone will match the VLAN ID strictly When others VLAN ID dismatch with us the packets will discard Contrarily the phone will accept the packets with the distinct VLAN ID 7 You must gain the IP with the Static mode when you set VLAN otherwise can t gain the IP in the VLAN and also can not dial with point to point 4 3 2 3 Service Port You can set the port of telnet HTTP RTP by this page NETWORK LOM LES service port 3 If modify HTTP or Telnet port you d better set it more than 1024 then restart SERVICE PORT set web browse port the default is 80 port if you want to HTTP Port enhance system safety you d better change it into non 80 standard port Example The IP address is 192 168 1 70 and the port value is 8090 the accessing address is http 192 168 1 70 8090 Telnet Port Set Telnet Port the default is 23 You can change the value into others Example The IP address is 192 168 1 70 the telnet port value is 8023 the accessing address is telnet 192 168 1 70 8023 RTP Initial Port Set the RTP Initial Port It is dynamic allocation RTP Port Quantity Set the maximum quantity of RTP Port the default is 200 Notice 1 You need save the configuration and reboot the phone after set this page 2 If you modify the port of Telnet and HTTP you would better set the value more than 1024 because the port value less than 1024 is system port reserved 3 if you set 0 for the HTTP port it will disabl
9. support 96ms e Support VAD CNG eo NAT transverse support STUN e Supports full duplex e SIP support SIP domain SIP authentication none basic MD5 DNS name of server peer to peer SIP support 2 servers user can through each server to calling in and out DTMF SIP info DTMF Relay RFC2833 e SIP application contain SIP call forward transfer holding waiting 3 way conference Paging and intercom click to dial pickup joincall redial unredial e Call control features Flexible dial map support hotline empty calling no reject server black list for reject authenticated call no disturb caller ID and so on Could dial use private server automatically when public server unregistered while private server is resgistered successfully Support phonebook 500 records incoming calls outgoing calls missing calls Each supports 100 records Support MWI support conference call in server Phonebook supports VCard standard Support 12 24 time format Support daylight saving time Support path gruu Support SIP Privacy 1 3 Network Features WAN LAN support Bridge mode Support PPPoE for xDSL support VLAN Support Stun penetration Support DHCP get IP on WAN port 0000000000000 45 Qos supports Diffserv support network tools contain ping trace route telnet client 1 4 Maintenance and Management The phone supports post mode can update firmware by post mode Supports different levels of administration Support Boot Monitor Can upgrade fir
10. which is encrypted or has some unique requirements or call flows Select DIMF sending mode there are three modes e DTMF RELAY 6 DTMF RFC2833 o DTMF_SIP_INFO Different VolP Service providers may provide different modes Select SIP protocol version to adapt for the SIP server which uses the same version as you select For example if the server is CISCO5300 you need to 27 R normally System uses RFC3261 as default Set Anonymous call out safely Support RFC3323and RFC3325 up Number Enable Displayname Set to make quotation mark to displayname as the phone sends out signal in order to be compatible with server Click to Talk Set click to Talk need practical software support Signal Encode Enable Disable Signal Encrypt RTP Encode Enable Disable RTP Encrypt Enable Timer Set Enable Disable Session Utne whether support ee a ee will refresh the SIP sessions Answer With Single Enable Disable the function when call is incoming Codec phone replies SIP message with just one codec which phone supports Auto Set to use O E TCP protocol to guarantee O E of transport as message is above 1300 byte Er atl the special SIP server when phone recieves the patckets sent from server phone will use the source IP address not the address in via field 4 3 3 2 Stun Config In this web page you can config SIP STUN STUN By STUN server the phone in private network could know the type of NAT and the NA
11. your desired number in phone book or callers Handfree Enter into hands free mode 1 4 Port for connecting POWER Powerswitch Select ON OFF Output 5V 1 0A Connect it to PC Connect it to Network The phone has two Network ports The WAN port and the LAN port Before you connect the power source please carefully read Safety Notices of this user manual 2 Initial connecting and Setting 2 1 connect the phone Step 1 Connect the IP Phone to the corporate IP telephony network Before you connect the phone to the network please check if your network can work normally You can do this in one of two ways depending on how your workspace is set up Direct network connection by this method you need at least one available Ethernet port in your workspace Use the Ethernet cable in the package to connect WAN port on the back of your phone to the Ethernet port in your workspace you can make direct network connect The following two figures are for your reference P Internet E gt D ADSL Cable Broadband Modem Router BW210 de internet c S l v ADSL Cable Modem BW210 Shared network connection Use this method if you have a single Ethernet port in your workspace with your desktop computer already connected to it First disconnect the Ethernet cable from the computer and attach it to the WAN port on the back of your phone Next use the Ethernet cable in the package to connect LAN port on the back of your phone t
12. 3 E A SN A sued vias AAA E AE A E A A A 24 A O a aa ak E 25 AI AO das 25 4332 O as 28 433 3 APESTA ee sae ete a Nee Re tee ee a acen sue 29 33 4 i PAONG A ee ee ee ee eee 32 4341 SP adas 32 EIA Gall SCE VICC TTT 33 43 43 Digital Map CONTIGUA ia 35 4 34 40 S feel BOOK a Ea AEAN 36 PS E A aicaiuacc sere coterie ees a a eae eee a a 37 K E TT Ce PEO A a id ia 37 43 92 ISO 0 CON Mica 38 K BE CONIO SENING A a EE E E E NARE 39 E R OG ALO EIA E EP S EA A T T E T A E SAE AES E E 39 4 9 9 9 AO cd 40 SOGE a 41 430 SOCUINY a E A ane ae ae eae 41 eR 1 MMI ATT celeste a a ee a 41 AIO A A ET Se Soe EN 42 437 ee Le A E E 43 4 4 SETTINGS VIA PHONE S KEYBOARD iia 44 4 4 1 How to set via the phone s kevboard ssssssssssssssssssssssssssssesens enean enen nenen 44 DAL NONE TIC HT suenccusuvasnedeovescueseenests 44 S APPENDI L aenee deere euueus eueeeaveeueamcg rune 45 5A SPECIFICATION rara a E A E douauaebeatiniesacuauubneesteueeauseunetans 45 5 4 1 DEVICES POCA UON iaa 45 Die VOICE FOUTE Sn E R L R 45 R R EP INCIWONGCT T sanini EE aladas 45 5 1 4 Maintenance and Management sssssssssssssssssssessssssesssseeesseeeee ennenen eenn neee eee 46 9 2DIGITSCHARACTER MAP TABLE oia aio ndi n 46 1 Introducing BW210 VoIP Phone 1 1 Thank you for your purchasing BW210 Thank you for your purchasing BW210 BW210 is a full feature telephone that provides voice communication over the same data network that your computer uses This phone
13. Answer Time Set Prefix in peer to peer IP call For example what you want to P2P IP Prefix dial is 192 168 1 119 If you define P2P IP Prefix as 192 168 1 you dial only 119 to reach 192 168 1 119 Default is If there is no Set it means to disable dialing IP MWI Number Set the number to listen voice mail in server Enable Call Transfer Enable Call Transfer by selecting it Enable Call Waiting Enable Call Waiting by selecting it Enable Three Way Enable Three Way Call Call Accept Any Call If select it the phone will accept the call even if the called number is not belong to the phone If select it the phone will auto answer when there is an incoming call Ban Outgoing If you select Ban Outgoing to enable it and you can not dial out any number Do Not Disturb Select NO Disturb the phone will reject any incoming call the callers will be reminded by busy but any outgoing call from the phone will work well Set Add Delete Black list If user does not want to answer Black List some phone calls add these phone numbers to the Black List and these calls will be rejected x and are wildcard x means matching any single digit for example 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out 34 DOT means matching any arbitrary number digit for example 6 expresses any number with prefix 6 will be forbidden to dialed out if user wants to allow a number o
14. Fanvil FANVIL TECHNOLOGY CO LTO anvi SMART VOIP Fanvil Product User Manual IP Phone Model BW210 Version V1 7 60 47 Mar 30 2009 2005 Fanvil technology Co Ltd All rights reserved This document is supplied by Fanvil Technology Co Ltd No part of this document may be reproduced republished or retransmitted in any form or by any means whatsoever whether electronically or mechanically including but not limited to by way of photocopying recording information recording or through retrieval systems without the express written permission of Fanvil Technology Co Ltd Fanvil Technology Co Ltd reserves the right to revise this document and make changes at any time and without the obligation to notify any person and or entity of such revisions and or changes Product specifications contained in this document are subject to change without notice W W W EA NY lL COM ADD Unit 4A Building NO 7 Tian An Industrial Park Nanshan District Shenzhen TEL 86 755 264 02199 Safety Notices Please read the following safety notices before installing or using this phone They are crucial for the safe and reliable operation of the device o Please use the external power supply that is included in the package Other power supplies may cause damage to the phone affect the behavior or induce noise o Before using the external power supply in the package please check with home power voltage Inaccurate power voltage may cau
15. Session Timer Answer With Single Codec Auto TCP Enable Strict Proxy Dial Without Register Ban Anonymous Call Enable GRUU APPLY i SIP Confic SIP Line Select SIP 1 lt Choose line to set info about SIP there are 2 lines to choose You can switch by Load button or not or so show Unapplied Phone Number Input the phone number assigned by your VolP service provider Phone will not register if there is no phone number configured Set proxy server IP address Usually Register SIP Server configuration is the same as Proxy SIP Server Proxy Server Address But if your VoIP service provider give different configurations between Register SIP Server and Proxy SIP Server you need make different settings Proxy Server Port Set your Proxy SIP server port Input your Proxy SIP server account Proxy Password Input your Proxy SIP server password Set the sip domain if needed otherwise this VolP phone will use the Register server address as sip domain automatically Usually it is same with registered server and proxy server IP address Enable Register Start to register or not by selecting it or not Domain Realm Register Expire Time Set expire time of SIP server register default is 60 seconds If the register time of the server requested is longer or shorter than the expire time set the phone will change automatically the time into the time NAT Keep Alive Interval User Agent Signal Key Medi
16. T mapping IP and port of SIP The phone might register itself to SIP server with global IP and port to realize the device both calling and being called in private network Stun server tell ant to receive data from 5060 Private Network Gateway NAT STUN Server 28 VOIP STUN Set sse T Set Sip Line Enable Stun SIP 1 v STUN STUN can penetrate NAT while False means not Set STUN Effective Time If NAT server finds that a NAT STUN Effect Time mapping is idle after time out it will release the mapping and the system need send a STUN packet to keep the mapping effective and alive Local SIP Port set the SIP port Set Sip Line Enable Stun Choose line to set info about SIP There are 2 lines to choose You can switch by Load button UseStun_____ Enable DisableSIPSTUN O O O Notice SIP STUN is used to realize SIP penetration to NAT If your phone configures STUN Server IP and Port default is 3478 and enable SIP Stun you can use the ordinary SIP Server to realize penetration to NAT 4 3 3 3 DIAL PEER setting This functionality offers you more flexible dial rule you can refer to the following content to know how to use this dial rule When you want to dial an IP address the entry of IP addresses is very cumbersome but by this functionality you can set number 156 to replace 192 168 1 119 here Number Destination port mode fatias Sufiw paienn ase roz1681119 5o60 ae noalis fno s
17. U FOR YOUR PURCHASING BW210 oooococccnccncconconconconconcnnnonnnnnnnrnnnn nr aaeeea aaaeeeaa 5 12 DELIVERY CONTENT e atai A T ade 5 VIS AREYPAD A aa E 5 LA PORT FOR CONNECTING auctio A A EE 7 2 INITIAL CONNECTING AND SETTING o ocoocoocconconconnonnnnnonnnnnnnnnn nono aaaeeeaa aeaaeae aeaaaee 7 2 1 CONNECT THE PHONE sss ss cesse eaea eaea aaaeeeaa aeaaeae aaeeea eaea ean 7 Pe IRA TTT io SE Oe Ae 8 22 E PPPOE MO on Ue 9 22 pies O Tee TTT 10 PRAES PE B u O SA le le asa 11 3 BASIC FUNG TIONS TTT 11 3 Le BASIC OPERATION srir ii ones 11 O Lee oo AAA gg A 11 A AAA gn A 11 E o AAA gn does sven ye vedas ceeevese eee 12 3 14 Transfering A a teeta 12 3 1 9 Calling Hold And 3 Ways A aaa 13 A A O EA NRT EY E A Ye 13 3 2 THE HIGH LEVELOPER ATI a 14 E ET O aad tena ato ae senor Aaa a a a aa aAa aa aa aaa aE 14 Bilis Gall PICKUP niana a A 14 3 2 3 E A A a a aa a Ei 15 3 2 4 POG Daaa E AEAEE A EAEE E OEE EEEE E I AEE ENE EEEE ENE SS EEEE S EE 15 SI E 6 TGT 15 Bh SETTING in 16 AA SETTING METHODS A A A A A SAA 16 AD SETTING VIA WEB BROWSE ui a 16 4 3 CONFIGURATION VIA WEB ci A A a e A e aa 17 AS TS BASIC aa 17 43 H E T A SenacuaegadbasnwSewebekvanac ivan danacutivasdtaued ducauusssebutaneannewareantusueeeavavassene 17 A TTT 17 ADA Be SR A E teh tusadyace E EA S couasaeeanneneceent 19 SS a L aE E a aa aaa E ania 19 JIZ UL y a O eS Dee Oe A a a Eee EE 20 4 32 WAN as 20 E SN A O CONMG APP Po O 21 AI UA O o A 2
18. a Key Local port Ring type Subscribe Expire Time Conference Number Enable DNS SRV Enable Subscribe Enable Keep Authentication NAT Keep Alive Enable Via rport Enable PRACK Long Contact Enable URI Convert Dial Without Register Ban Anonymous Call Forward Type Forward Phone Number Server Type DTMF Mode RFC Protocol Edition recommended by the server and register again Set examining interval of the server default is 60 seconds Enable Disable Keep Authentication Enable Disable keeps NAT of SIP alive If some server refuse to register with too short interval time and has no packets sending to device in private network to keep NAT alive user could set this function ON It need set the keep alive interval time less than the NAT server s Enable Disable system to support RFC3581 Via rport is special way to realize SIP NAT default config SEM server Convert to 23 when send the URI Set call out by proxy without registration Set to ban Anonymous Call Select call forward mode the default is Off Off Close down calling forward Busy If the phone is busy incoming calls will be forwarded to the appointed phone No answer If there is no answer incoming calls will be forwarded to the appointed phone Always Incoming calls will be forwarded to the appoint phone directly The phone will Prompt the incoming while doing forward Appoint your forward phone number Select the special type of server
19. al to B 4 Setting 4 1 Setting methods VoIP Phone is different from the traditional phone it need be set to make it active If your VolP service provider asks you to set this phone you can do it easily according to the following methods This VolP Phone can be set via three different setting methods The phone key The initial password is 123 for setting via phone key The web browser on PC Telnet This Manual will tell you about the setting methods via the web browser on PC 4 2 Setting via Web Browse When this phone and your PC are connected to your network enter the IP address of the wan port in this phone as the URL e g http xxx xxx xxx xxx or http Xxx XXX XXX XXX XXXX If you do not know the IP address you can look it up on the phone s display by pressing the key SYSINFO After you enter the IP address you will see the following web interface Username Password This phone provides different two privileges for different users to set it The two privileges are guest and administrator respectively In guest privilege user can see but not modify Register Proxy Sever Addresses and ports of SIP advance SIP and lax2 In administrator privilege user can see and modify all setting parameters Default value in guest privilege Username guest Password guest Default value in Administrator privilege Username admin Password admin Input username and password click logon and you wi
20. an add or delete user account and change the authority of each user account in this web page MAINTENANCE A Ec Marista Mi ACCOUNT Mareen Set Keyboard Password Keyboard Password User Set Account Option Account Configuration Keyboard Password Set the password for entering the setting menu of the phone by the phone s key board The password is digit 40 mame et This table shows the current user existed User Name Set account user name sss configuration General can only read Password Confirm the password Select the account and click the Modify to modify the selected account and click the Delete to delete the selected account General user only can add the user whose level is General 4 3 5 6 Reboot MAINTENANCE PON Mac Ma MU asta REBOOT Reboot Phone Press the Reboot button to reboot Phone If you modified some configurations which need the phone s reboot to be effective you need click the Reboot then the phone will reboot immediately Notice Before reboot you need confirm that you have saved all configurations 4 3 6 Security 4 3 6 1 MMI Filter SECURITY MMI FILTER BS MMI Filter Table MMI Filter Table Set po fra MMI Filter Table Set User could make some device own IP which is pre specified access to the MMI of the phone to config and manage the phone MMI Fileter IP Table list Start IP 41 MMI Filter Table Set Add or de
21. appointed prefix code After making the above configuration A can dial 2 plus B or C number to join B and C s call User can set prefix in random in the case of no affecting current dialing rules 3 2 4 redial unredial If B is in busy line when A calls B A will get notice busy please hang up If A want to connect B as soon as B is in idle he can use redial function at the moment and he can dials an appointed prefix number plus B s number to realize redial function What is redial function A can t not build a call with B when B is in busy then A will subscribe B s calling mode at 60 second intervals once B is available A will get reminder of rings to hook off while A hooks off A will call B automatically If at this time A is occupied temporarily and unwilling to contact B A also can cancel the redial function by dialing an appointed prefix plus B s number before making the redial function Number T AKT 3 is appointed prefix code After making the above configuration A can dial 3 plus B phone number to make the redial function 4 is appointed prefix code After configuration A can dial 4 to cancel redial function User can set prefix in random in the case of no affecting current dialing rules 3 2 5 click to dial When user A browses in an appointed Web page user A can click to call user B via a link this link to user B then user A s phone will ring after A hooks off the phone will di
22. e HTTP service 4 3 2 4 SNTP Setting time zone and SNTP Simple Network Time Protocol server according to your location you can also manually adjust date and time in this web page NETWORK TOO eer ME a SNTP SNTP Time Set Time Zone GMT 08 00 Beijing Chongging Hong Kong Urumqi Ka 12 Hours Systems O OOO S Daylight Timeset Mor SNTP 12 Hours Systems Swich the time mechanism between 12 hours and 24 hours Default is 24 hours mode 24 Enable Daylight Time shift minutes ay Setup start and end day Setup start and end day O S Setup start and end hours SNTP Select the SNTP and click Apply to make the SNTP Times effective Setup start and end minutes Notice You need specify the above all items 4 3 3 VOIP 4 3 3 1 SIP Config Set your SIP server in the following interface VOIP sip REO MN gt SIP Line Select stp 1 y Basic Setting Proxy Server Address 192 168 1 2 Proxy Server Port 50 E Phone Number i 4 25 Register Expire Time seconds NAT Keep Alive Interval seconds User Agent Voip Phone 1 0 i Key ort i pe S M L R 1 U Type 102 subscribe Expire Time ag seconds gnal dia K ng Ty Conference Number Enable Keep Authentication Forward Phone Number A ee EN RFC Privacy Edition EA Transfer Expire Time Enable Conference Number Enable Displayname Quote Enable
23. eans any phone No that starts with your set phone number will be sent via SIP2 line after the first several digits of your dialed phone number are deleted according to delete length Phone Number Destination optional Port optional Alias optional Call Mode Suffix optional ST 4 3 4 Phone 4 3 4 1 DSP Config This setting will realize speed dial function after you dialing the numeric key 2 the number after all will be sent out The phone will automatically send out alias number adding your dialed number if your dialed number starts with your set phone number You need set Phone Number Alias and Delete Length Phone number is XXXT and Alias is Rep xxx If your dialed phone number starts with your set phone number the first digits same as your set phone number will be replaced by the alias number specified and New phone number will be send out If your dialed phone number starts with your set phone number The phone will send out your dialed phone number adding suffix number When you dial 2 the SIP1 server will receive 33334444 When you dial 8309 the SIP1 server will receive 07558309 When you dial 0106228 the SIP1 server will receive 86106228 When you dial 147 the SIP1 server will receive 1470011 In this page you can configure voice codec input output volume and so on 32 PHONE per OaS SIA AUS SS DSP Configurati
24. er for those parameters 2 3 Press the Press the key the LCD screen will display INPUT PASSWORD Input password default is 123 then press the gt key the LCD screen will display NETWORK Enter Press the key and the LCD screen will display LAN Press the key then Enter the ss key the LCD screen will display STATIC NET Enter key the LCD screen will display IP Press the key again and then the key input your desired IP address for your IP phone and confirmed by pressing the key then the LCD will display the inputted IP address When inputting IP with keypad use instead of Exit gt Press the key to return to previous menu then press the key the LCD Enter screen will display DNS2 Press the S then the key input your spare DNS address and confirm it by pressing the the inputted DNS address Exit key and then the LCD will display Press the key to return to previous menu then press the key the LCD Enter screen will display DNS Press the namasa key then the key input your DNS Enter address and confirm it by pressing the gt key and then the LCD will display the inputted DNS address Exit gt Press the key to return to the previous menu and then press the key the Enter LCD screen will display GATEWAY Press the gt key again and then the Enter key input yo
25. ess for auto update 37 The address can be IP address or Domain name with subdirectory Set FTP server Username System will use anonymous if username keep blank Password Set FTP server Password Config File Name Set configuration file s name which need to update System will use MAC as config file name if config file name keep blank For example 000102030405 Config Encrypt Key Input the Encrypt Key if the configuration file is encrypted Protocol Type Select the Protocol type FTP TFTP or HTTP Update Interval Time Set update interval time unit is hour Different update modes 1 Disable means no update Update Mode 2 Update after reboot means update after reboot 3 Update at time interval means periodic update 4 3 5 2 Syslog Config Syslog is a protocol which is used to record the log messages with client server mechanism Syslog server receives the messages from clients and classifies them based on priority and type Then these messages will be written into log by some rules which administrator can configure This is a better way for log management 3 levels in debug information Level 0 emergency This is highest default debug info level You system can not work Level 1 alert Your system has deadly problem Level 2 critical Your system has serious problem Level 3 error The error will affect your system working Level 4 warning There are some potential dangers But your system can w
26. et delete length This is optional config item For example if the delete length is 3 the phone will delete the first 3 digits then send out the rest digits You can refer to examples of different alias application to know how to set delete length Introduction of how to set up dial peer to implement switch between multi SIP Ines T 8T or Rann foso lsm hoaias aans o 9T mapping If you have registered a SIP1 server and set dial peer according to the above table all calls will be sent via SIP1 server when you press the numeric key 9 in front of dialing destination phone numbers BT mapping If you have registered a Private SIP2 server and set dial peer according to the above table all calls will be sent via SIP2 server when you press the numeric key 8 in front of dialing destination phone numbers Number Destination Port Mode Alias Suffix Del length 2T 0 0 0 0 4569 IAX2 del no suffix 1 the rule of 2T means user need to dial the number with prefix 2 if he want to dial via I AX2 server Examples of different alias application Set by web explanation example You need set phone If you dial number Destination Alias 93333 the SIP2 festiantion opilonal and Delete Length server will receive Port optional Phone number is XXXT 3333 S Destination is Call Mode SIP 255 255 255 255 and Alias Suffix optional is del Delete Length optional f1 This m
27. from port 1 switch will transmit it to port 2 the other port in the red VLAN and not transmit it to port3 and port 4 in blue VLAN By this means VLAN divide the broadcast domain via restricting the range of broadcast frame transmition Note chart 2 use red and blue to identify the different VLAN but in practice VLAN uses different VLAN IDs to identify VLAN 2 AG Broadcast Domain After Switchboard received i the Broadcast Frame only transmit it to other H k port which belong to same VLAN with send port X d X Chan 2 NETWORK We oos ESC ME LC VLAN Enable VLAN ID Check Enable Voice Data VLAN differentiated Undifferentiated Voice 802 1P Priority lo 0 7 Data 302 1P Priority lo 0 7 4095 Data VLAN ID 254 0 4095 QoS Configuration VLAN Enable Before select it to enable VLAN you need enable Bridge mode in LAN config Enable VLAN ID check by selecting it After enable VLAN VLAN ID Check ID check if VLAN ID of a data package is not the same Enable with the phone s or a data package do not have VLAN ID the data package will be discarded After enable VLAN system will set packets with different type of VLAN ID Undifferentiated means after using VLAN both VoIP packets and other data packets will use 22 Voice Data VLAN the voice VLAN ID tag differentiated means after using differentiated VLAN VolP signal and voice packets will add voice VLAN ID and other data packets wi
28. functions not only much like a traditional phone allowing to place and receive calls and enjoy other features that traditional phone has but also it own many data services features which you could not expect from a traditional telephone This guide will help you easily use the various features and services available on your phone 1 2 Delivery Content Please check whether the delivery contains the following parts The base unit with display and keypad The handset The handset cable The power supply The Ethernet cable 1 3 Keypad Sysinfo The numeric keypad with the keys 0 to 9 and is used to enter Digits and letters additionally the following keys are available R Send Key mapping Ke Key y name c Menu Phone Book gt Callers O LED Sysinfo System amish Informatio n Confirm Navigation Key aC lt gt MWI Transfer Transfer Conferen ce Conference Delete O Hold O Function Description In idle state press the MENU key to call up the menu In idle mode press the Phone Book key to check the record list and add new records and revise the record Press this key again will return to idle mode In idle pickup calling mode press the Callers key to Check the Income Outgoing Missed calls records Press this key again will return to idle mode LED blinks to remind user new voicemail In idle mode press the Sysinfo key to check the phone setting parameter
29. g you need input your desired phone number in Phone number this blank and then you need dial the phone number to realize calling to what the phone number is mapped In the prefix matching you need input your desired prefix number and T then dial the prefix and a phone number to realize calling to what your prefix number is mapped The prefix number supports at most 30 digits Set Destination address This is optional config item If Destination you want to set peer to peer call please input destination IP address or domain name If you want to use this dial 30 OS rule in SIP2 line you need input 255 255 255 255 or 0 0 0 2 in it gt Port SettheSignal port the default is 5060 for SIP A Set alias This is optional config item If you don t set Alias it will show no alias Note There are four types of aliases 1 add xxx it means that you need dial xxx in front of phone number which will reduce dialing number length 2 all xxx it means that xxx will replace some phone number 3 del It means that phone will delete the number with length appointed 4 Rep It means that phone will replace the number with length and number appointed You can refer to the following examples of different alias application to know more how to use different aliases and this dial rule Call Mode Select differenct signal protocol SIP or IAX2 Set suffix this is optional config item It will show no suffix if you don t set it Delete Length S
30. he last page Like Static IP MODE Choose PPPoE MODE click NEXT to config the PPPoE account password and SIP default SIP1 You can browse it too Click BACK to return to the last page Like Static IP MODE PPPOE Set PPPOE Server ANY Username ImputyourADSLaccount 23232 Notice Click Finish button after finish your setting IP Phone will save the setting automatically and reboot After reboot you can dial by the SIP account 4 3 1 3 Call Log You can look up all the outgoing calls through this page BASIC aO BE ca o Call information Start Time Lastrime called Number Call Log 4 3 1 4 MMI SET BASIC ESAS Pues come MATES MMI SET LANGUAGE SELECTION Greeting Message Set Greeting Message VOIP PHONE Version VOIP PHONE V1 7 60 47 Mar 30 2009 16 39 26 MMI SET explanation Language Set Set the language of phone English is default 4 3 2 Network 4 3 2 1 WAN Config NETWORK J wan oos ad E WAN Status MAC Address 00 0e 10 00 66 10 WAN Setting Static O DHCP PPPOE APPLY WAN Config Field Name Active IP 192 168 1 23 Current Netmask 255 255 255 0 Current Gateway 192 168 1 1 MAC Address 00 0e 10 00 66 10 The current IP address of the phone The current Netmask address The current MAC address of the phone The current Gateway IP address Shows the time of getting MAC address WAN Setting Static O Please select the proper network mode according to the net
31. lete the IP address segments that access to the phone Set initial IP address in the Start IP column Set end IP address in the End IP column and click Add to add this IP segment You can also click Delete to delete the selected IP segment Select it or not to enable or disable MMI Filter Click Apply to make it effective Notice Do not set your visiting IP outside the MMI filter range otherwise you can not logon through the web 4 3 6 2 Firewall SECURITY ES mear NM Firewall Type L In_access Enable Ul Out_access Enable APPLY Hrewall Input Rule Table Index beny Permi protocolsrc addr Srcmask Des Addr pes Mask Range Jort Firewall Output Rule Table Index Deny Permilprotocolsrc Addr srcmask DesAddr Des Mask Range port Firewall Set Deny protocol Type uon v re mask PortRange Tana v Rule Delete Input outout Input indexa peves T Firewall Configuration In this web interface you can set up firewall to prevent unauthorized Internet users from accessing private networks connected to the Internet input rule or prevent unauthorized private network devices from accessing the Internet output rule Firewall supports two types of rules input_access rule and output_access rule Each type supports at most 10 items Through this web page you could set up and enable disable firewall with input output rules System could prevent unauthorized access or access other networks set in rules for security
32. ll add data VLAN ID data untaged means after using VLAN only VoIP packets will add voice VLAN ID Other data packets will not use VLAN DiffServ Enable DiffServ Value Voice 802 1P Priority Data 802 1P Priority Set 802 1p of data VLAN Non VolP data such as http telnet ping etc will use this value to set VLAN package Voice VLAN ID Data VLAN ID Set 802 1q of data VLAN ID Non VolIP data such as http telnet ping etc will use this value to set VLAN package NOTICE 1 Startup VLAN if set Voice Data VLAN differentiated as Undifferentiated all packets will use the Voice VLAN ID as the tag 2 Startup VLAN if set Voice Data VLAN differentiated as tag differentiated and disable the DiffServ then system will not distinguish the voice and data all packets will use the Voice VLAN ID as the tag 3 Startup VLAN if set Voice Data VLAN differentiated as tag differentiated and enable the DiffServ then system will distinguish the voice and data and add the VLAN ID each other 4 Startup VLAN if set Voice Data VLAN differentiated as data untaged then the packet of the signal voice will use the Voice VLAN ID as the tag but the data packets will not take the VLAN tag 5 If Disable the VLAN regardless to set the Voice Data VLAN differentiated or not all packets will not take the VLAN tag If enable the DiffServ all packets will only take the DiffServ value 6 user need notice enable the VLAN ID Check Enable that is default If enable
33. ll enter setting web interface There is a selection menu on the left side of the web interface Click on the desired submenu the current settings of this submenu will be displayed in the larger field on the right You can now modify and store the values by using mouse and keyboard of your PC To save the changes click on the submenu maintenance and then click the config button and the Save button on the right field 4 3 Configuration via WEB 4 3 1 BASIC 4 3 1 1 Status BASIC status RUZ MAMA ES SS Network Phone Number SIP LINE 1 5060 Unapplied SIP LINE 2 5060 Unapplied Version VOIP PHONE V1 7 60 47 Mar 30 2009 16 39 26 Status Field name Explanation Shows the configuration information on WAN and LAN port including the connect mode of WAN port Static DHCP PPPoE MAC address the IP address of WAN port and LAN port ON or OFF of DHCP mode of LAN port Phone Number Shows the phone numbers provided by the SIP LINE 1 2 servers The last line shows the version number and issued date 4 3 1 2 Wizard BASIC F errs wizero RMR rear ser Network Mode Select Field Name Explanation PPPoE MODE Please select the proper network mode according to the network condition FV6030 provide three different network settings e Static If your ISP server provides you the static IP address please select this mode then finish Static Mode setting If you don t know about parameters of
34. mware through boot monitor access with different authority support auto provisioning Can config through Web Keypad Telnet Can upgrade firmware and configuration file through HTTP FTP TFTP Support syslog 00000000900 5 2 Digit character map table Button Character Button Character Y 2ABCabc 9 aw fl 8TUVtuv S S or v BN 6 uno 46
35. n If Tn is not specified it is assumed to be TO by default on all dial plans RULE 1 8 xxx 99T4 9911x T4 1 8 xxx Cause extensions 1000 8999 to be dialed immediately 9xxxxxxx Cause 8 digit numbers started with 9 to be dialed immediately 911 Cause 911 to be dialed immediately after it is entered 99T4 Cause 99 to be dialed after 4 seconds 9911x T4 Cause any number started with 9911 to be dialed 4 seconds after dialing ceases Notice End with Fixed Length Time out and Digital Map Table can be used simultaneously System will stop dialing and send number according to your set rules 4 3 4 4 Phone Book You can input the name phone number and select ring type for each name here 36 Phonebook Table Index Add Phone Book Phone Book Shows the detail of current phonebook Name Shows the name corresponding to the phone number Number Shows the phone number Ring Type Shows the ring type of the incoming call Click Modify to change the selected information and click the Delete to delete the selected record Notice the maximum capability of the phonebook is 500 items 4 3 5 Maintenance 4 3 5 1 Auto Provision MAINTENANCE AUTO PROVISION ETS Mel MO MR Reed S Auto Update Setting Update Interval Time Update Mode APPLY Current Config Show the current config file s version Version Server Address Set FTP TFTP HTTP server IP addr
36. nd 2 are attended transfer 3 is blind transfer Notice to VolP Phone Carrier Your VolP phone server need support FRC3515 or else transferring can not work 3 1 5 Calling Hold and 3 ways call There are two modes to enjoy hold function 1 Press the key during a call and the call will be on hold While a call is on hold you can establish another call by dialing your desired number and confirm it by the button Pressing the key again you will resume the first call By using hold function you can talk with only one party the other party who is on hold can t talk with you If you press the button or Key you will enter into 3 ways call 2 If the third party calls you during a call the LCD screen will display the incoming call number Press the key to hold the first call and then you can talk with the third party By using hold function you can talk with only one party the other party who is on hold can t talk with you If you press key phone will hang up the first call and then accept the new incoming call Notice You must enable the calling waiting or else calling hold can t work 3 1 6 Callers The VoIP phone maintains lists of missed received and dialed calls Each list can contain up to 100 entries If the call list capacity is full new call will replace the first call If you stop power supply or restart the phone the record will disappear Missed Calls Press the T and then the 1W Y4 key till
37. ngth of dialing In order to keep some users secondary dialing manner when dialing the external line with pbx phone can be added a special rule to realize it so user can dial a number as external line prefix and get the secondary dial tone to keep dial the external number after finishing dialing phone will send the prefix and external number totaly to ther server for example there is a rule 9 xxxxxxxx in the digital map table after dialing 9 phone will send the secondary dial tone user may keep going dialing after finished phone will call the number which starts with 9 actually the number sent out is 9 digit with 9 Ja PHONE Mos cani service erra map BN O eaen NN E Digital Rule table Digital Map Configuration Field name explanation Set Enable Disable the phone ended with dial Endwith Fixed Length Specify the Fixed Length of phone ending with Set the timeout of the last dial digit The call will be sent Time out after timeout Digital Rule table Below is user defined digital map rule Specifies a range that will match digit May be a range a list of ranges separated by commas or a list of digits x Match any single digit that is dialed Match any arbitrary number of digits including none Tn Indicates an additional time out period before digits are sent of n seconds in length n is mandatory and can have a value of 0 to 9 seconds Tn must be the last 2 characters of a dial pla
38. o your desktop computer Your IP Phone now shares a network connection with your computer The following figure is for your reference ADSL Cable Modem BW210 E fenis t Step 2 Connect the handset to the handset port by the handset cable in the package Step 3 connect the power supply plug to the DC port on the back of the phone Use the power cable to connect the power supply to a standard power outlet in your workspace Step 4 push the on off switch on the back of the phone to the on side then the phone s LCD screen displays WAIT LOGON Later a ready screen typically displays the date time and current network mode If your LCD screen displays different information from the above you need refer to the next section Initial setting to set your network online mode If your VoIP phone registers into corporate IP telephony Server your phone is ready to use 2 2 Initial Setting This VolP Phone provides you with rich function and parameters setting If you have enough knowledge about network and SIP protocol it is better for you to understand many parameters But if you know little about network and SIP protocol you can also easily make initial setting according to the following steps to enjoy rapidly high quality voice and low cost from this VolP Phone Before make initial setting please check if your corporate IP telephony network can work 8 normally and you have finished connect the phone
39. on Output volume 5 15 E DE DSP Configuration explanation The fist preferential DSP codec G 711A u G 722 G 723 G 729 G 726 The second preferential DSP codec G 711A u G 722 G 723 G729 G 726 The third preferential DSP codec G 711A u G 722 G 723 G 729 G 726 The forth preferential DSP codec G 711A u G 722 G 723 G 729 g 726 The fifth preferential DSP codec G 711A u G 722 G 723 G 729 G 726 The sixth preferential DSP codec G 711A u G 722 G 723 G 729 G 726 Input Volume Specify Input MIC Volume grade Handfree Volume Specify Handfree Volume grade Set G729 Payload Length Length Specify the least reflection time of Handdown the default is 200ms Specify Output receiver Volume grade Specify Ring Volume grade 160 20ms or 320 20ms is available 5 3kb s or 6 3kb s is available Set up the ring by default Select Signal Standard VAD Select it or not to enable or disable VAD If enable VAD G729 Payload length could not be set over 20ms 4 3 4 2 Call Service In this web page you can configure Hotline Call Transfer Call Waiting 3 Ways Call Black List white list Limit List and so on 33 PHONE PEA cali service ME MUSEE SO Call Service Setting puto answer O T TEH bonoro IS AN S Black List a Limit List Limit List po fa fx ________ lala Call Service A Specify Hotline number If you set the number you can not dial any other numbers Specify No
40. ork Level 5 notice Your system works well in special condition but you need to check its working environment and parameter Level 6 info the daily debugging info Level 7 debug the lowest debug info Professional debugging info from R amp D person At present the lowest level of debug information send to Syslog is info debug level only can be displayed on telnet MAINTENANCE REA SYSLOG Ma MU wis a Reed a Syslog Set MGR Log Level None Syslog Configuration 38 Enable Syslog Select it or not to enable or disable syslog 4 3 5 3 Config Setting MAINTENANCE NIE Mtra CONFIG ME Mr Reed Save Configuration Press the Save button to save the configuration files Backup Configuration Save all Network and VoIP settings Right Click here to Save as Config File txt Clear Configuration Press the Clear button to Clear the configuration files Config Setting you can save all changes of configurations Click the Save Config Save button all changes of configuration will be saved and be effective immediately Right clicks on Right click here and select Save Target As then you will save the config file in txt format user can restore factory default configuration and reboot the phone Clear Config lf you login as Admin the phone will reset all configurations and restore factory default if you login as Guest the phone will reset all configurations except fo
41. r VoIP accounts SIP1 2 and IAX2 and version number 4 3 5 4 Update You can update your configuration with your config file in this web page MAINTENANCE a MES Mesta UPDATE Ted MT Web Update Select miel BB fp ba au Update FTP Update Application update FIP APPLY Update explanation Click the browse button find out the config file saved 39 Web Update before or provided by manufacturer download it to the phone directly press Update to save You can also update downloaded update file logo picture ring mmiset file by web Server Set the FTP TFTP server address for download upload The address can be IP address or Domain name with subdirectory Password name is the MAC of the phone such as 000102030405 Notice You can modify the exported config file And you can also download config file which includes several modules that need to be imported For example you can download a config file just keep with SIP module After reboot other modules of system still use previous setting and are not lost Action type that system want to execute 1 Application update download system update file 2 Config file export Upload the config file to FTP TFTP server name and save it 3 Config fie import Download the config file to phone from FTP TFTP server The configuration will be effective after the phone is reset Protocol Select FTP TFTP server 4 3 5 5 Account Config You c
42. r a series of number incoming he may add the number s to the list as the white list rule the configuration rule is number for example 123456 or 1234xx Black List 4119 Means any incoming number is forbidden except for 4119 Note End with DOT when set up the white list Set Add Delete Limit List Please input the prefix of those Limit List phone numbers which you forbid the phone to dial out For example if you want to forbid those phones of 001 as prefix to be dialed out you need input 001 in the blank of limit list and then you can not dial out any phone number whose prefix is 001 x and are wildcard x means matching any single digit for example 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out means matching any arbitrary number digit for example 6 expresses any number with prefix 6 will be forbidden to dialed out Notice Black List and Limit List can record at most10 items respectively 4 3 4 3 Digital Map Configuration This phone supports 4 dial modes 1 End with dial your desired number and then press 2 Fixed Length the phone will intersect the number according to your specified length 3 Time Out After you stop dialing and waiting time out system will send the number collected 4 User defined you can customize digital map rules to make dialing more flexible It is realized by defining the prefix of phone number and number le
43. s Such as local phone number local IP and local Gateway IP address Use the Enter key to enter next menu or confirm the setting Use the Exit key to return to previous menu cancel the setting or reject to answer a call When you pick up the handset or during calling you can use this key to turn up or turn down the handset volume when a call comes you can use this key to adjust ring volume you also can use this key to choose item in the menu callers or phone book Use this key to read old or new message Use the key to realize blind transfer or attended transfer please refer to 3 1 4 call transfer for more details Use this key to realize the three party call pleae refer to3 1 5 Calling Hold and 3 ways call for more details In menu use this key to modify current setting or delete invalid information And when you input number do call out you can use it to delete characters Temporarily hold the active call during the talking press the key again to unhold the call You also can press this key then input the third party s phone number and end with the key during calling you can make a call with the third party and hold the previous calling 3 1 5 Calling Hold and 3 ways call Press this key in calling mode you can hear the other side and the other side can not hear you In the hook off hands free mode use the key to R send dial the last call number use this key to make a quick dial as soon as you select
44. se fire and damage Please do not damage the power cord If power cord or plug is impaired do not use it it may cause fire or electric shock e The plug socket combination must be accessible at all times because it serves as the main disconnecting device e Do not drop knock or shake it Rough handling can break internal circuit boards e Do not install the device in places where there is direct sunlight Also do not put the device on carpets or cushions It may cause fire or breakdown e Avoid exposure the phone to high temperature below 0 C or high humidity Avoid wetting the unit with any liquid e Do not attempt to open it Non expert handling of the device could damage it Consult your authorized dealer for help or else it may cause fire electric shock and breakdown e Do not use harsh chemicals cleaning solvents or strong detergents to clean it Wipe it with a soft cloth that has been slightly dampened in a mild soap and water solution e When lightning do not touch power plug or phone line it may cause an electric shock e Do not install this phone in an ill ventilated place e You are in a situation that could cause bodily injury Before you work on any equipment be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents Table of Content 1 INTRODUCING BW210 VOIP PHONE 2 css sss ss ss ssc sse seene aaeeea eaea aaaeeeaa 5 1 1 THANK YO
45. splay PASSWORD Then press the ENTER key and the DEL key input Enter your PPPoE s password and confirm it by the the password which you inputted 7 Press the EXIT key for four times and press the DOWN key till the LCD screen display SYSTEM Key the LCD screen will display Enter 8 Press the ENTER key the screen display SAVE then press the key again the LCD screen will display ARE YOU SURE E nter 9 Press the key the phone will save your setting and the LCD screen will display SAVING NOW then return to display SAVE Exit 10 Press the key twice then a numeric key 3 and hold until the screen display ARE YOU SURE Press the key the screen will display CHANGING which means that the phone is trying to switch to PPPoE mode If the icom PPPoE on the top of the screen keeps blink it shows that the 1 Is trying to access the PPPoE server and the IP is still static IP if you press key to display the current IP if the icon PPPoE is showed without blink it means that the phone has already gotten IP from PPPoE server 2 2 2 Static IP mode le Enter Press the 1 key for three seconds then confirm it by the key your phone network connection mode will switch into Static IP mode Prepare your phone s network parameters They are IP Address of this phone Subnet Mask Default Gateway Router and DNS You can ask your VolP service provid
46. the LCD screen display MISSED Press the key the LCD screen will display the missed call number 13 and sequence numbersof the missed call R Send You can press the key to dial this phone number also press UP DOWN key to browse the other missed calls or you can press the key again the LCD screen will display the time of the missed calls If there is no one missed calls the LCD will display LIST IS EMPTY eo Received Calls ZTN Press the key and then the Y A till the LCD screen display RECEIVED Press ENTER key the LCD screen will display the received call number and sequence number of the received call JN You can press the key to dial this phone number you also B DO key to browse the other received calls or you can press the key again the LCD screen will show the time of the received call If there is no one received call the LCD will display LIST IS EMPTY o Dialed calls Press the T and then the 1W Y4 key till the LCD screen display OUTGOING Press key the LCD screen will display the phone number and R Send sequence number of the dialed call You can press the key to dial this phone L number or press the k key to browse all record of the dialed calls If there is on one dialed calls the LCD will display LIST IS EMPTY 3 2 The high level operation This VolP Phone provides more advanced functions after setting at the permission scope of SIP server 3 2
47. umn to logon in the phone 3 If networks ID which is distributed by DHCP server is same as network ID which is used by LAN of system phone will use the DHCP IP to set WAN and modify LAN s networks ID for example system will change LAN IP from 192 168 10 1 to 192 168 11 1 when phone uses DHCP client to get IP in startup if phone uses DHCP client to get IP in running status and network ID is also same as LAN s phone will refuse to accept the IP to configure WAN 4 3 2 2 Qos Config The VOIP phone support 802 1Q P protocol and DiffServ configuration VLAN functionality can use different VLAN IDs by setting signal voice VLAN and data VLAN The VLAN application of this phone is very flexible Do not use VLAN After Switchboard received the Broadcast Frame transmit to every other port except the send port Switchboard Broadcast Frame In Chart 1 21 Use VLAN Switchboard e A A E J av 4 A P 4 F f Broadcast Frame P Broadcast Frame VLAN 1 J A A Domain H Ny b T In chart 1 there is a layer 2 switch without setting VLAN Any broadcast frame will be transmitted to the other ports except the send port For example a broadcast information is sent out from port 1 then transmitted to port 2 3and 4 In chart 2 red and blue indicate two different VLANs in the switch and port 1 and port 2 belong to red VLAN port 3 and port 4 belong to blue VLAN If a broadcast frame is sent out
48. ur gateway s IP address and confirm it by pressing the key the LCD screen will display the inputted gateway address gt a Press the key to return to the previous menu and then press the key the Enter LCD screen will display NETMASK Press the key again and then the Enter key input your netmask and press the key to confirm it The LCD screen will display the inputted netmask a 10 Press the key for four times and press the key till the LCD Screen displays SYSTEM Enter Enter 11 Press the key the LCD screen will display save then press the key again the LCD screen will display ARE YOU SURE Enter 12 Press the SAVE key this phone will display SAVING NOW then return to display Exit 13 Press the o gt key twice to exit the menu and then na the numeric key 1 till the LCD screen displays ARE YOU SURE Press the key the LCD screen will display CHANGING If the icon static on the top of screen shows without blink it means phone has already used the static IP 2 2 3 DHCP mode Press the numeric key 2 and hold till the LCD screen displays ARE YOU SURE Press the Enter key the LCD screen will display CHANGING and this VoIP phone is trying to switch to DHCP mode If the icom DHCP on the top of the screen keeps blink it shows that Sysinfo the phone is trying to access the DHCP server and the IP is
49. utis o When you want to dial a long distance call to Beijing you need dial an area code 010 before local phone number but you can also dial number 1 instead of 010 after we make a setting according to this dial rule For example you want to dial 01062213123 but you need dial only 162213123 to realize your long distance call after you make this setting 1T 1 0 0 0 0 5060 SIP rep 010 no suffix 29 To save the memory and avoid abundant input of user add the follow fuctions umber enman port mode alas sufix amn me Rann 5060 am maa mans R aisan jowoo 5o60 se judio no sutis jo 1 x Match any single digit that is dialed If user makes the above configuration after user dials 11 digit numbers started with 13 the phone will send out 0 plus the dialed numbers automatically 2 Specifies a range that will match digit It may be a range a list of ranges separated by commas or a list of digits If user makes the above configuration after user dials 11 digit numbers started with from 135 to 139 the phone will send out 0 plus the dialed numbers automatically Use this phone you can realize dialing out via different lines without switch in web interface VOIP SA Bettie DIAL PEER Dial Peer Table Add Dial Peer Destination optional Delete Length optional Dial Peer Option DIAL PEER There are two types of matching conditions one is full matching the other is prefix matching In the Full matchin
50. work condition FV6030 provide three different network settings eo Static If your ISP server provides you the static IP address please select this mode then finish Static Mode setting If you don t know about parameters of Static Mode setting please ask your ISP for them DHCP In this mode you will get the information from the DHCP server automatically need not to input this information artificially PPPoE In this mode your must input your ADSL account and password You can also refer to 3 2 1 Network setting to speed setting your network tmask f you use static mode you need set it IP Address Input the IP address distributed to you Netmask Input the Netmask distributed to you Input the Gateway address distributed to you Set DNS domain postfix When the domain which you DNS Domain inputted can not be parsed phone will automatically add this domain to the end of the domain which you inputted before and parse it again Alter DNS Input your standby DNS server address PPPOE Server If you uses PPPoE mode you need to make the above setting PPPoE Server It will be provided by ISP Input your ADSL account Password Input your ADSL password Notice 1 Click Apply button after finishe your setting IP Phone will save the setting automatically and new setting will take effect 2 If you modify IP address the web will not response by the old IP address Your need input new IP address in the address col
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