Fanvil Product User Manual IP Phone Model: BW206
Contents
1. tone 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 2 The high level operation This VoIP Phone provides more advanced functions after setting at the permission scope of SIP server 3 2 1 Special Keys O Realize Secondary Dial by Dialing for only one time rm When you make secondary dial in off hook handsfree mode press Sit key to postpone input One hold stands for 2 seconds For example you input 123 45 the phone will send DTMEF 45 2 seconds after the phone call 123 123 45 will make phone send DIMF 45 at 6 seconds interval 3 2 2 Call pickup Call pickup is implemented by simulating pickup function of PBX it s that when A calls B B 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 o 0000 foso fsm frepmicap fno sutt 3 1 means appointed prefix code After making the above configuration C can dial 1 plus the phone number of B to pick up A s call User can set prefix in random in the2. 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 will 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 STATUS MANO AMS ES I _ Phone Number SIP LINE 1 5060 Unapplied SIP LINE 2 T 5060 Unapplied Version VOIP PHONE V1 7 346 141 Explanation Shows the configuration information on WAN and LAN port Network 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 system version 4 3 1 2 Wizard BASIC RE wizero A E gt Network Mode Select A AAA bemos O AA EE R Field Name Explanation PPPoE MODE Please select the proper network mode according to the network condition BW2
3. NAT Keep Alive Interval Set the user agent if have the default is VoIP Phone 1 0 Enable Keep Enable Disable Keep Authentication Authentication Enable Disable keeps NAT of SIP alive NAT Keep 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 Via rport Enable Disable system to support RFC3581 Via rport is special a way to realize SIP NAT Enable PRACK Enable or disable SIP PRACK function suggest use the default server Enable URI Convert Convert to 23 when send the URI Dial Without Register Set call out by proxy without registration Ban Anonymous Call Set to ban Anonymous Call Select call forward mode the default is Off Off Close down calling forward Forward Type 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 Forward Phone Number Appoint your forward phone number 22 Server Type Select the special type of server which is encrypted or has some unique requirements or call flows Select DTMF sending mode there are three modes DTMF_RELAY
4. 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 8309 the SIPI server will receive 07558309 When you dial 0106228 the SIP1 server will receive 86106228 When you dial 147 the SIP1 server will receive 1470011 4 3 4 Phone 4 3 4 1 DSP Config In this page you can configure voice codec input output volume and so on PHONE psp MAME A A DSP Configuration Output Volume 5 1 9 2 DSP Configuration explanation The fist preferential DSP codec G 711A u G 722 G 723 G 729 G 726 Second Codec The second preferential DSP codec G 711A u G722 G723 G 729 G 726 Third Codec The third preferential DSP codec G 711A u G722 G723 G 729 G 726 Forth Codec The forth preferential DSP codec G 711A u G 722 G 723 G 729 8 726 Fifth Codec The fifth preferential DSP codec G 711A u G 722 G 723 G 729 G 726 Sixth Codec The sixth preferential DSP codec G 711A u G722 G723 G 729 G 726 Input Volume Specify Input MIC Volume grade G729 Payload Length Set G729 Payload Length H
5. fno suts fo 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 8T 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 Examples of different alias application Set by web explanation example You need set phone number If you dial 93333 Destination Alias and Delete the SIP2 server will Length receive 3333 Phone number is XXXT Destination is 255 255 255 255 CallMode SIP and Alias is del This means 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 This setting will realize speed When you dial 2 dial function after you dialing the SIP1 server will Port optional the numeric key 2 the number receive 33334444 Alias optional all 33334444 after all will be sent out Call Mode SIP Y Suffix optional AAA Delete Length optional DoS Phone Number Destination optional The phone will automatically send out alias number adding your dialed number if your dialed number starts with your set phone number
6. prevent unauthorized access or access other networks set in rules for security 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 Input Output mo rend S ProtacolType ue v en PS Port range moremo pesn Field name explanation Select it to Enable in_ access rule Select it to Enable out_ access rule Input Output Specify current adding rule by selecting input rule or output rule Specify current adding rule by selecting Deny rule or Permit rule Protocol Type Filter protocol type You can select TCP UDP ICMP or IP eos Set source address It 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 Src Mask 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 Des Mask 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 Out
7. 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 mn 1 When A talks to B B may press the Sis key and dial C phone number After B talks to C or B Trans hear alert from C B presses the key then B hangs up and A will get through to C 2 When Ais 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 will get through to C key dial C phone number and key then hang up and A 1 and 2 are attended transfer 3 is blind transfer Notice to VoIP Phone Carrier Your VoIP 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 Pp 1 Press the Be 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 rm Hold 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 you will enter into 3 ways call 2 If the third party calls you during a call the top led would blink and the phone would paly call waiting
8. 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 Number Destination Port Mode Alias Suffix Del Length To save the memory and avoid abundant input of user add the follow functions 24 1 x Match any single digit that 1s dialed If user makes the above configuration after user dials 11 digit numbers started with 13 the phone will send out O 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 O plus the dialed numbers automatically Use this phone you can realize dialing out via different lines without switch in web interface VOIP SON Beanie DIAL PEER Dial Peer Table Dial Peer Option Po AL PEER PEER Field name There are two types of matching conditions one is full matching the other 1s prefix matching In the Full matching you need input your desired phone number in this blank and then you need dial the Phone number 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 w
9. 06 provide three different network settings Oo Static If your ISP server provides you the static IP address please select this mode and then finish Static Mode setting If you don t know about parameters of 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 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 Static IP Address 192 168 1 179 192 168 1 1 onspomain NN Static IP Address Input the IP address distributed to you Input the Netmask distributed to you Input the Gateway address 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 Primary DNS Input your primary DNS server address Alter DNS Input your standby DNS server address SIMPLE SIP SET User Mame po o 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 Register on Display detailed informa
10. DTMF Mode Oo DTMF_RFC2833 DTMEF_SIP_INFO Different VoIP Service providers may provide different modes Select SIP protocol version to adapt for the SIP server which uses RFC Protocol Edition the same version as you select For example if the server is CISCO5300 you need to change to RFC2543 else phone may not cancel call normally System uses RFC3261 as default Transport Protocol Set transport protocols TCP or UDP RFC Privacy Edition Set Anonymous call out safely Support RFC3323and RFC3325 Transfer Expire Time The phone send bye and end the call as soon as hang up Number Quote out signal in order to be compatible with server Signal Encode Enable Disable Signal Encrypt RTP Encode Enable Disable RTP Encrypt Enable Session Timer Set Enable Disable Session Timer whether support RFC4028 It will refresh the SIP sessions Answer With Single Codec Enable Disable the function when call is incoming phone replies SIP message with just one codec which phone supports a transport as message is above 1300 byte Enable Strict Proxy Support the special SIP server when phone receives the pickets sent from server phone will use the source IP address not the address in via field Enable GRUU Set to support GRUU 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 NAT mapping IP and port of SIP The phone might register itself t
11. MAC address of the phone The current Gateway IP address Shows the time of getting MAC address WAN Setting Static Please select the proper network mode according to the network condition FV6030 provide three different network settings o Static If your ISP server provides you the static IP address please select this mode and 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 Static IP Address 192 168 1 179 DNS Domain 70256154133 02 56 1028 68 If you use static mode you need set it Input the IP address distributed to you Input the Netmask distributed to you Input the Gateway address distributed to you Set DNS domain postfix When the domain which you inputted can DNS Domain not be parsed phone will automatically add this domain to the end of the domain which you inputted before and parse it again Primary DNS Input your primary DNS server address Alter DNS Input your standby DNS server address Obtain DNS server automatically Select it to use DHCP mode to get DNS address If you disable it you will use static DNS server The default is enabling it PPP
12. MAS O fearen oo Timeowt S80 Digital Rule table Set Enable Disable the phone ended with FF dial Set the timeout of the last dial digit The call will be sent after Time out 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 plan 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 OXxxxxxx Cause 8 digit numbers started with 9 to be dialed immediately 30 911 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 FUNCTION KEY Configuration This phone supports 10 memory keys for speed dial You could save 10 numbers from Fl to F10 Then you could lift ha
13. OE Server NY Username user123 Password e If you uses PPPoE mode you need to make the above setting PPPoE Server It will be provided by ISP Input your ADSL account Input your ADSL password Notice 1 Click Apply button after finished 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 column 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 16 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 switchboard Aga uE _ __ the Broadcast Frame transmit to every other port except the send port Broadcast Frame Chart 1 Use VLAN After Switchboard received ciety a the Br
14. RTP Initial Port Itis 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 O for the HTTP port it will disable 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 19 NETWORK TOO MOE ME aa SNTP SNTP Time Set sever oe GMT 08 00 Beijing Chongging Hong Kong Urumqi A2 Hours systems O O OOOO Daylight Timeset Field name Server Time Zone Time Out Default is 24 hours mode effective Enable Daylight Time shift minutes Month Week ay Setup start and end day Hour Setup start and end hours Minute 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 20 VOIP SIP REO Mor SIP Line Select SIP 1 as Proxy Server Address Sanane __ ioza6sa2 2 E Account Name Proxy Password gi stat Display Name Advanced SIP Setting Register Expire
15. TV FANVIL TECHNOLOGY CO LTO anv SMART VOIP Fanvil Product User Manual IP Phone Model BW206 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 WWW FANVIL 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 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 cause fire and damage o Please do not damage t
16. Time 60 seconds __ rorward tye om User Agen vopPronet0 server twe common gt Media Key RFC3261 v Local Port SIP Config SIP Line Select Ea ss AAA Choose line to set info about SIP there are 2 lines to choose You can switch by Load button Register Status Shows if the phone has been registered the SIP server or not or so show Unapplied 21 Phone Number Input the phone number assigned by your VoIP service provider Phone will not register if there 1s no phone number configured Display Name Set the display name Set proxy server IP address Usually Register SIP Server configuration is the same as Proxy SIP Server But if your VoIP Proxy Server Address 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 VoIP phone will use Domain Realm the Register server address as sip domain automatically Usually it is same with registered server and proxy server IP address Set expire time of SIP server register default is 60 seconds If the Register Expire Time 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 recommended by the server and register again
17. anddown Time Specify the least reflection time of Handdown the default is 200ms Handfree Volume Specify Handfree Volume grade VA Select it or not to enable or disable VAD If enable VAD G729 Payload length could not be set over 20ms Dtmf payload type Set up DTMF payload type 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 PHONE MSN CALL SERVICE MM MS Hotline PS papi Presi oe se APPLY T T 28 Call Service Specify Hotline number If you set the number you can not dial any other numbers Specify No Answer Time Set Prefix in peer to peer IP call For example what you want to dial is 192 168 1 119 If you define P2P IP Prefix as 192 168 1 you dial only 119 P2P IP Prefix to reach 192 168 1 119 Default is If there is no Set it means to disable dialing IP 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 Auto handdown Auto Handdown Time Do Not Disturb Black List Limit L
18. ave Target As then you will save the config file in txt format User can restore factory default configuration and reboot the phone If you login as Admin the phone will reset all configurations and Clear Config restore factory default if you login as Guest the phone will reset all configurations except for VoIP accounts SIP1 2 and version number 4 3 5 4 Update You can update your configuration with your config file in this web page MAINTENANCE MN Ma Mata UPDATE Ma A Web Update Select file M z ba au l_ Update FTP Update Server Username Password File Name A Type Protocol P APPLY Field name Click the browse button find out the config file saved before or Web Update provided by manufacturer download it to the phone directly press Update to save You can also update downloaded update file ring mmiset file by web Server Set the FTP TFTP server address for download upload The address 33 o can be IP address or Domain name with subdirectory 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 sy
19. ble Set MMI Filter Table Set C MMI Filter APPLY _ MMI Filter 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 MMI Filter Table Set Add or delete 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 35 MMI Filter 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 FIREWALL nar o ooo Hrewall Type C Out_access Enable Firewall Input Rule Table A ai wala wae O deny ICMP 192 168 1 14 255 255 255 0 192 168 1 118 255 255 255 i por range e Index To e veretes OO 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
20. 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 inputting 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 umber Destination Port Mode Janas sutis 2 means 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 wants 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 3 1s appointed prefix code After making the above configuration A can dial 3 plus B s phone number to make the redial function 4 is appointed prefix c
21. ccccccccceccccccccccccccccccces 4 EZ DELIVERY CONTENT A ces usuebeedescivessoetests 4 PLEASE CHECK WHETHER THE DELIVERY CONTAINS THE FOLLOWING PARTS 4 THE BASE UNIT WITH KEYPAD cuisine iii 4 TAE ENDS ED aii EE 4 THE HANDSET CABLE nas 4 THE POVE RSU PP coins o 4 THE ETHERNET CABLE siini a r T E O O 4 MD CYP ND EE EEE ORI A E E EE EEE FA EE ETEA EE E E E PETA EE E IEE A E AA EE TA 4 KEY MAPPING haia a al 5 La PORTS HOR CONNECTING cnoiar eee a a a 5 2 INITIAL CONNECTING AND SETTING constan iia 6 Ze CONNE CF THE PHONE aer a aa aa aa aer SE ESENES 6 22A NITIAE SETTING oscense ea aeaa r a S ia 7 Ss BASIC EUING TIONS y a AA A AEA 8 3 12 BASIG OPERA TION amp sesccccccscsctsavescuccccdeccicscvessceeccedlecdsasseseovencdeaddecscssosoedbodeceedevansVacsoesseudavcousccessvessseucevecdenacbessecss 8 DPA EA COCONINO A aio 8 3 L2 MAKE A A A A a ea a iaeaea 8 O QUICK DIALING omiiia eni E A E T 8 Dh FENCING ACA a A A A a tusieduveg sti e aaua 8 DLA STARSTETIA LACA AAA A A A A i a aaaea 9 3 1 3 Calling Holdand 3 ways Calla ia A AA A AA A a iaae sesen 9 3 2 THE HIGH LEVEL OPERA TION aiii ii 9 32 Y 1 E K yS eannan a a a e a aa Gate ede eaten 10 A AA NS 10 EA AAA A eh a tee eae cece aaa a aa rotate ie O T 10 324 TOGIGI ANT COTA AAA cahoots hice heen aa aaa a a aaa a a eae 10 ES E UT OIG AA A A E Il ES LIN Gian vcotecckclescssovacrecetecedonctvonsecohesadercecmese cont ondcvercocsmechoiaderenaceracceosstoneneeulareneiasone
22. e 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 the voice VLAN ID tag Voice Data VLAN differentiated means after using VLAN VolP signal and voice differentiated packets will add voice VLAN ID and other data packets will add data VLAN ID data untagged means after using VLAN only VoIP packets will add voice VLAN ID Other data packets will not use VLAN DiffServ Enable Select it or not to Enable or disable DiffServ DiffServ Value Set DiffServ value the common value is 0x00 Voice 802 1P Priority Specify 802 1P Priority of voice signal data package etc will use this value to set VLAN package 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 disables 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 enables 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 untagged then the packet of the signal voice will us
23. e 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 it the phone will match the VLAN ID strictly When others VLAN ID doesn t match 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 PRU LES service port B RTP Initial Port RTP Port Quantity 1200 APPLY 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 1f you want to enhance HTTP Port 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 1s 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 1s telnet 192 168 1 70 8023 RTP Initial Port Set the
24. erence Paging and intercom click to dial pickup joincall redial unredial Call control features Flexible dial map support hotline empty calling no reject server black list for reject authenticated call no disturb and so on 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 Qos supports Diffserv support network tools contain ping trace route telnet client v o o 5 1 4 Maintenance and Management The phone supports post mode can update firmware by post mode Supports different levels of administration Can upgrade firmware through boot monitor access with different authority support auto provisioning Can config through Web Telnet Can upgrade firmware and configuration file through HTTP FTP TFTP Support syslog
25. ewselnientcaicassuontacenciateniseesteese 12 AT SETTING METHODS it a 12 42 SETTING VIA WEB BROWSE ia 12 43 CONFIGURATION VISA WEB ia 13 LRL BASICO A A A ED ISITE ADE NIE Ser ne tase OA MRE REA Aor aR RIC NG T tre erE EEO eR RT 13 SN CL W o 15 4 ES PT VOI AA NN IR 20 7 S E 11 N ERE PAREAN PEET TA VENEA AANE SETA AN EEEN E VENEET EE beau EETA ET 27 4o Man DU CO aaa a EP AN 31 DIAS COC UTN is 35 SR O 37 SAP PENDEA a TA A A seeecceeiaas a easars aa 38 BU SPECIFICATION aeai eo 38 1 Introducing BW206 VoIP Phone 1 1 Thank you for your purchasing BW206 Thank you for your purchasing BW206 BW206 is a full feature telephone that provides voice communication over the same data network that your computer uses This phone functions not only much like a traditional phone allowing to place and receive calls and enjoy other features that traditional phone has but also 1t 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 keypad The handset The handset cable The power supply The Ethernet cable 1 3 Keypad The numeric keypad with the keys 0 to 9 and is used to enterDigits and letters additionally the following keys are available a wi a po oe Key mapping Func
26. g 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 8 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 work 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 for 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 ONE SYSLOG Meese MN PR Musee O Syslog Set Syslog Configuration 32 4 3 5 3 Config Setting MAINTENANCE NOE Mtra CONFIG MiNi MM Save Configuration Press the Save button to save the configuration files Backup Configuration You can save all changes of configurations Click the Save button Save Config all changes of configuration will be saved and be effective immediately Backup Config Right clicks on Right click here and select S
27. git 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 29 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 length 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 totally to their 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 PHONE me MARES DIGITAL MAP
28. hat your prefix number is mapped The prefix number supports at most 30 digits Set Destination address This is optional config item If you want to Destination set peer to peer call please input destination IP address or domain name If you want to use this dial rule in SIP2 line you need input 255 255 255 255 or 0 0 0 2 in it Set alias This is optional config item If you don t set Alias it will show no alias 25 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 different signal protocol SIP Suffix Set suffix this is optional config item It will show no suffix if you don t set it Delete Length Set 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 lines er ooo2 500 stp foals
29. he power cord If power cord or plug is impaired do not use it it may cause fire or electric shock The plug socket combination must be accessible at all times because it serves as the main disconnecting device Oo Do not drop knock or shake it Rough handling can break internal circuit boards eo 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 Avoid exposure the phone to high temperature below 0 C or high humidity Avoid wetting the unit with any liquid eo 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 O 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 BW 206 VOIP PHONE isiiiinis aaa 4 1 1 THANK YOU FOR YOUR PURCHASING B W200 ccccccccccccccccccccccccccccccccccccccccccccccccccc
30. ist The phone will hang up and return to standby automatically at hands free mode After this time the phone will hang up and return to standby automatically at hands free mode 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 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 DOT means matching any arbitrary number digit for example 6 expresses any number with prefix 6 will be forbidden to dial out If user wants to allow a number or 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 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 di
31. ndset and press Fn number to dial the number directly PHONE Interface Configuration MWI Number 4 3 5 Maintenance 4 3 5 1 Auto Provision MAINTENANCE AUTO PROVISION EEx Ma MU e Misco Auto Update Setting Current Config Version Server Address Username Config Encrypt Key 5 ai Update Interval Time Update Mode Disable Auto Provision Current Config Version Show the current config file s version Server Address Set FTP TFTP HTTP server IP address for auto update The address can be IP address or Domain name with subdirectory Username Set FTP server Username System will use anonymous if username 31 keep blank 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 Enable DHCP Option 66 If this option is enabled TFTP server address defaults to the value of option 66 4 3 5 2 Syslog Config Syslog is a protocol which is used to record the log messages with client server mechanism Syslo
32. ng calls E Press the button O If you need switch from a hands free call to handset please pick up the handset directly O If you need switch from a handset call to hands free please press the up the handset button and then hang 3 1 2 Making a call O Quick dialing In idle mode input the called number and press key or ISEE button phone will dial the call and use hands free automatically O Use handset Pick up the handset and you will hear dialing tone right now Then input the phone number and end by the or button When you hear ringback tone du du from handset the call is through After talking hang up the handset to end the call O Use hands free Press the button and you will hear dialing tone at the same time Then input the phone number and end by the or EE button When you hear ringback tone du du from handset the call is through After talking press button to end the call t Oo Use the Redial key Please pick up handset or press the key After you hear dialing tone please press the ESSE key to dial the last called number Note after you reboot the phone the phone will clear the redial record so there is no redial number 3 1 3 Ending a call O Hangs up by handset on hook o when in hands free O Hangs up by press O 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
33. o SIP server with global IP and port to realize the device both calling and being called in private network Stun server tell ustomer public etwork IP and 2345 port AT apping port 12345 ant to receive data from 5060 Public Network Gateway NAT STUN Server 23 penetrate NAT while False means not Set STUN Effective Time If NAT server finds that a NAT mapping STUN Effect Time 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 7 Local SIP Port _ Set the SIP port Set Sip Line Enable Stun SIP 1 v Choose line to set info about SIP There are 2 lines to choose You can switch by Load button Enable Disable SIP STUN 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 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 instead of 010 after
34. oadcast Frame only transmit it to other port which belong to same VLAN with send port J j Broadcast Broadcast Frame Frame VLAN VLAN 2 WG gt Broadcast Ly Broadcast gt a e Domain Y Domain MS In chart 1 there 1s a layer 2 switches without setting VLAN Any broadcast frame will be transmitted to the other ports except the send port For example a broadcast information 1s 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 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 Chan 2 17 NETWORK EVON cos ESC Mine VLAN ID Check Enable Voice Data VLAN differentiated Undifferentiated Voice 802 1P Priority o O 7 Data 302 1P Priority 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 ID VLAN ID Check Enable check if VLAN ID of a data package is not the same with the phone s or a data package do not hav
35. ode 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 10 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 dial to B 11 4 Setting 4 1 Setting methods VoIP Phone is different from the traditional phone it need be set to make it active If your VoIP service provider asks you to set this phone you can do it easily according to the following methods This VoIP Phone can be set via three different setting methods The web browser on PC Telnet This part 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 Kxx xxx XXX XXX XXXX If you do not know the IP address you can look it up by IVR of Local IP inquiry After you enter the IP address you will see the following web interface Username Password Logon 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 ports of SIP and advance SIP In administrator privilege
36. ource 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 wt Internet gt gt SS ADSL Cable Broadband Modem Router r temet S ADSL Cable Modem 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 to your desktop computer Your IP Phone now shares a network connection with your computer The following figure is for your reference J Internet ADSL Cable Modem 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
37. 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 LED would be lit Soon it would be off until system starts up Then it would be lit again If your VoIP phone registers into corporate IP telephony Server your phone is ready to use 2 2 Initial Setting This VoIP 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 VoIP Phone Before make initial setting please check if your corporate IP telephony network can work normally and you have finished connect the phone 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 VoIP Phone s network connection setting 3 Basic Functions 3 1 Basic operation 3 1 1 Accepting a call There are four methods to accept an incoming call O Pick up handset to accept incomi
38. put 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 192 168 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 Click the Delete button to delete the selected rule 4 3 7 Logout System Logout Click Logout and you will exit web page If you want to enter it next time you need input user name and password again 37 5 Appendix 5 1 Specification 5 1 1 Device specification this VoIP Phone Adapter Input Output Input 100 240VAC 50 60Hz Output 5SV 1A 10 100Base T RJ 45 for LAN Auto MDIX LAN 10 100Base T RJ 45 for PC Auto MDIX Idle 1 5W Active 1 8W Operation Temperature 0 40 C Relative Humidity 10 65 Main Chipset SDRAM Flash 2Mbits Size WxHxD 11 6x8x3 in Q95x205x75mm 07 094 5 1 2 Voice Features Support 2 lines SIP SIP 2 0 RFC3261 Codec G 711A u G7231 high low G729 G722 G726 Echo cancellation Support G 168 and hand free can support 96ms Support VAD CNG NAT transverse support STUN Supports full duplex 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 DTME SIP info DTMF Relay RFC2833 SIP application contain SIP call forward transfer holding waiting 3 way conf
39. stem 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 Select FIP TFTP server 4 3 5 5 Account Config You can add or delete user account and change the authority of each user account in this web page MAINTENANCE A Ec Meelis Milo wim ACCOUNT MT Set Keyboard Password Keyboard Password fe Keyboard Password Set the password for entering the setting menu of the phone by the Ii prone igy tor The passwords ig Root Genera This table shows the current user existed Set user level Root user has the right to modify configuration 34 General can only read Set the 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 Maca Mail MU wi Maat REBOOT Reboot Phone Reboot 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 MS MMI Filter Table MMI Filter Ta
40. tion Deseription Local IP Press speaker and then press the key you would hear the human voice with phone s active IP address Local IP Local Numer Press speaker and then press the key you would hear the human voice with phone s SIP phone number Local NUM lt gt Release During talking by handset pressing the key would let you close the current call and get new dial tone Rlease Hold Temporarily hold the active call during the talking press p the key again to resume the call You can also press this key then input the third party s phone number and end Hold with the key during calling and then you can make a call with the third party and hold the previous calling JP Transfer Use the key to do blind transfer or attended transfer Trans Mute Press this key during talking you can hear the other side but the other side could not hear you Mute vol Volume control Adjust the ring volume and talking voice volume VOL O a 10 mmber or or saved 10 number for fast dialing Send Press this key to make a quick dial as soon as you select ian your desired number in phone book or callers or send EZ the number you dialed manually po Redial In the hook off hands free mode use the key to dial the last call number Enter into hands free mode 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 s
41. tion 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 the 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 PPPoE Server It will be provided by ISP Input your ADSL account Input your ADSL password 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 MESES BO cau Los BM Call information Start Time last Time Called Number 4 3 1 4 MMI SET BASIC B status FE wizaro SUS MM SET III Language Selection Version VOIP PHONE V1 7 346 141 MMI SET Language Set Set the language of phone English is default 4 3 2 Network 4 3 2 1 WAN Config NETWORK wan REN EE E CC I 4 4 S WAN Status Current Netmask 255 255 255 0 o 192 168 1 1 00 01 0e 61 00 98 Get MAC Ti 20110419 WAN Setting Obtain DNS server automatically APPLY WAN Config Field Name 15 Active IP 192 168 1 23 Current Netmask 255 255 255 0 MAC Address 00 0e 10 00 66 10 The current IP address of the phone The current Netmask address The current
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