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fanvil product user manual i p phone model: bw2 06 ? 2005 fanvil technology co,. ltd all rights reserve d. this document is supplied by fanvil t echnology 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 ph otocopying, recording, information recording or through retrieval systems, without the express written permission of fanvil t echnology 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 speci?cations contained in this document are subject to change without notice. free datasheet http://
2 safety notices please read the following safety notices before installing or using this phone. they are crucial for the safe and reliable operatio n 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. ? before using the external power supply in the package, please check with home power voltage. inaccurate power voltage may cause 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. ? the plug - socket combination must be accessible at all times b ecause it serves as the main disconnecting device. ? do not drop, knock or shake it. rough handling can break internal circuit boards. ? 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 ? do not attempt to open it. non - expert hand l ing of the device could damage it. consult your authorized dealer fo r help, or else it may cause fire, electric shock and breakdown. ? 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. ? when lightning, do not touch power plug or phone line, it may cause an electric shock. ? do not install this phone in an ill - ventilated place. ? you are in a situation that could cause bodily injury. before you work on any equipment, be aware of the hazards involved with e lectrical circuitry and be familiar with standard practices for preventing accidents. free datasheet http://
3 table of content 1. introducing BW206 vo ip phone ................................ ................................ ................................ ...... 4 1.1. t hank you for your pu rchasing BW206 ................................ ................................ ............................... 4 1.2. d elivery c ontent ................................ ................................ ................................ ................................ ...... 4 please check whether the delivery contain s t he following parts: ...................... 4 the base unit with k eypad ................................ ................................ ................................ ....................... 4 the handset ................................ ................................ ................................ ................................ ....................... 4 the handset cable ................................ ................................ ................................ ................................ ........ 4 the power supply ................................ ................................ ................................ ................................ ........... 4 the ethernet cable ................................ ................................ ................................ ................................ ..... 4 1.3. k eypad ................................ ................................ ................................ ................................ .......................... 4 key mapping: ................................ ................................ ................................ ................................ ...................... 5 1.4. p orts for connecting ................................ ................................ ................................ ............................... 5 2. initial connecting and setti ng ................................ ................................ ................................ ....... 6 2.1. c onnect the phone ................................ ................................ ................................ ................................ .... 6 2.2. i nitial s etting ................................ ................................ ................................ ................................ ............ 7 3. basic functions ................................ ................................ ................................ ................................ ........... 8 3.1. b asic operation ................................ ................................ ................................ ................................ .......... 8 3.1.1. accepting a call ................................ ................................ ................................ ................................ ....... 8 3.1.2. making a call ................................ ................................ ................................ ................................ .......... 8 ? q uick - dialing ................................ ................................ ................................ ................................ ............ 8 3.1.3. ending a call ................................ ................................ ................................ ................................ ........... 8 3.1.4. transferring a call ................................ ................................ ................................ ................................ .. 9 3.1.5. calling hold and 3 ways call ................................ ................................ ................................ .................. 9 3.2. t he high - level operation ................................ ................................ ................................ ........................ 9 3.2.1. special keys ................................ ................................ ................................ ................................ ........... 10 3.2.2. call pickup ................................ ................................ ................................ ................................ ............ 10 3.2.3. join call ................................ ................................ ................................ ................................ ................. 10 3.2.4. redial/unredial ................................ ................................ ................................ ................................ ....... 10 3.2.5. click to dial ................................ ................................ ................................ ................................ ........... 11 4. setting ................................ ................................ ................................ ................................ ............................ 12 4.1. s etting methods ................................ ................................ ................................ ................................ ....... 12 4.2. s etting via w eb b rowse ................................ ................................ ................................ ......................... 12 4.3. c onfiguration via web ................................ ................................ ................................ .......................... 13 4.3.1. basic ................................ ................................ ................................ ................................ ................... 13 4.3.2. network ................................ ................................ ................................ ................................ ................. 15 4.3.3. voip ................................ ................................ ................................ ................................ ...................... 20 4.3.4. phone ................................ ................................ ................................ ................................ ..................... 27 4.3.5. maintenance ................................ ................................ ................................ ................................ .......... 31 4.3.6. security ................................ ................................ ................................ ................................ .................. 35 4.3.7. logout ................................ ................................ ................................ ................................ ................... 37 5. appendix ................................ ................................ ................................ ................................ ......................... 38 5.1. s pecification ................................ ................................ ................................ ................................ ............. 38 free datasheet http://
4 1. introducing BW206 voip phone 1.1. thank you for your purchasing BW206 thank you for your p urchasing 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 featu res 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 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, a dditionally, the following keys are available: free datasheet http://
5 key mapping: key key name function description local ip p ress speaker, and then press the key, you would hear the human voice with phone t ransfer use the key to do blind transfer or attended transfer . m ute p ress this key during talking, you can hear the other side , but the other sid e could not hear you. v olume control adjust the ring volume and talking voice volume memory key t here are 10 memory keys(or called speed dial keys) saved 10 number for fast dialing. s end press this key to make a quick dial as soon as you select your desired number in phone book or callers , or send the number you dialed manually. redial in the hook off /hands - free mode, use the key to dial the last cal l number; handfree enter into hands - free mode. 1.4. port s for connecting power power switch select on/off dc power port output: 5v/ 1.0 a lan network port connect it to pc wan network port connect it to network t he 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. free datasheet http://
2. initial connecting and setting 2.1. connect the phone step 1: connect the ip phone to the corporate i p 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 w an 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 referenc e. 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 y our 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 . 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: pus h the on/off switch on the back of the phone to the on side, then the phones l e d 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 . free datasheet http://
2.2. initial setting this voip phone provide s you with rich function and parameters setting. if you ha ve enough knowledge about network and sip protocol, it is better for you to understand many parameters. but if you know little about networ k 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. b efore make i nitial setting, please check if your corporate ip telephony network can work normally, and you have finished connect the phone. t his 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 do es nt support dhcp, you need change this voip phone s network connection setting. free datasheet http://
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 . ? press the b u tton . ? if you need switch from a hands - free call to handset, please pick up the handset directly. ? if you need switch from a handset call to hands - free, pleas e press the button, and then hang up the handset. 3 .1.2. making a call ? quick - dialing in idle mode, i nput the called number, and press # key or button , phone will dial the call and use hands - free automatically . ? use handset pick u p the handset, and you will hear dialing tone right now . t hen input the phone number and end by the # or but ton. when you hear ringback tone du, du from handset , the call is through. a fter talking, h ang up the ha ndset to end the call. ? use hands - free press the button and you will hear dialing t one at the same time . t hen input the phone number and end by the # or button. when you hear ringback tone du, du from handset , the call is through. a fter talking, press button to end the call. t ? use the r edial key please pick up handset or press the key. after you hear dialing tone, please press the 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 ? hangs up by handset on hook ? hangs up by press when in hands - free free datasheet http://
9 ? 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 c all transfer has s everal 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 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 key, dial c phone number and # key, then hang up and a will get through to c. 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 en joy hold function : 1 . press the key during a call, and the call will be on h o ld. while a call is on h o ld, 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 cant 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 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 h old cant 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 cant work. 3.2. the high - level operation t his voip phone provide s more advanced functions after setting at the permission scope of sip server. free datasheet http://
10 3.2.1. special keys ? realize secondary dial by dialing for only one time when you make secondary dial in off - hook/handsfree mode, press key to postpone input . one hold ( -- ) 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. 3.2.2. call pickup call pickup is implemented by simula ting pickup function of pbx . its that, when a call s b, b rings but no answer, at this moment, c can hook off and input an appointed prefix plus bs number, pick up as 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 the phone number of b to pick up a s call. u ser can set prefix in random, in the case of no affecting current dialing rul es . 3 . 2 . 3. join call w hen 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 jo in call function. * 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. u ser can set prefix in random, in the case of no affecting current dialing rules . 3.2.4. redial/unredial i f b is in busy line when a calls b , a will get notice: busy, please hang up. if a want s 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 bs number to realize redial function. w hat is redial function ? a can t not build a call with b when b is in busy, then a will subscribe bs calling mode at 60 second intervals . once b is available , a will get reminder of rings to hook off, while a hook s off, a will call b automatically . i f 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 bs number before making the redial function. * 3 * is 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 pref i x code . after configuration, a can dial *4* to cancel redial function. u ser can set prefix in random, in the case of no affecting current dialing rules . free datasheet http://
11 3.2.5. clic k to dial when user a browse s in an appointed web page , user a can click to call user b via a link (this link to user b), then user as phone will ring, after a hook s off, the phone will dial to b . free datasheet http://
4 . setting 4 .1. s etting 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: t he web bro wser 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://xxx.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. this phone provides different two privil eges 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 address es , port s of sip and advance sip. in administrator privilege, user can see and modi fy all set ting 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 fiel d 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 s ave button on the right field. free datasheet http://
4.3. configuration via web 4.3.1. basic 4.3.1.1. status field name explanation networ k 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 provide d by the sip line 1 - 2 servers. the last line shows the system version. 4.3.1.2. wizard wizard field name explanation please select the proper network mode according to the network condition. BW206 provide three different network settings: ? static: if your isp server provides you the static ip address, please select this mode, and then finish static mode setting. if you dont 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 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. free datasheet http://
14 static ip address input the ip address distributed to you. netmask input the netmask distributed to you. gateway input the gateway address distributed to you. dns domain set dns domain postfix. when the domain which you input ted can not be parsed, phone will automatically add this domain to the end of the domain which you input ted before and parse it a gain. primary dns input your primary dns server address. alter dns input your standby dns server address. display name if user s et the display name , callee will show this display name. server address input your sip server address. server port set your sip server port. user name input your sip register account name. password input your sip register password. phone number input the phone number assigned by your voip service provider. enable register s tart to register or not by selecting it or not . 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 the last page. l ike static ip mode choose pppoe mode click next to config the pppoe account/password and sip(default sip1) . you can browse it to o. click back to return to the last page. l ike static ip mode pppoe server it will be provided by isp. username input your adsl account. free datasheet http://
15 password input your adsl password. notice: click finish button after finish your setting, ip phone will save th e setting automatically and reboot . after reboot, you can dial by the sip accou n t. 4.3.1.3. call log you can look up all the outgoing calls through this page. call log field name explanation start time display the start time of the outgoing call las t time display the conversation time of the outgoing call . called number display the account/protocol/line of the outgoing call . 4.3.1.4. mmi set mmi set field name explanation language set set the language of phone, english is default. 4.3.2. n etwork 4.3.2.1. wan config wan config field name explanation free datasheet http://
16 active ip the current ip address of the phone. current netmask the current netmask address. mac address the current mac address of the phone. current gateway the cur rent gateway ip address. get mac time s hows the time of getting mac address please select the proper network mode according to the network condition. fv6030 provide three different network settings: ? static: if your isp server provides you the static i p address, please select this mode, and then finish static mode setting. if you dont 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. if you use static mode, you need set it. ip addr ess input the ip address distributed to you. netmask input the netmask distributed to you. gateway input the gateway address distributed to you. dns domain set dns domain postfix. when the domain which you input ted can not be parsed, phone will automat ically add this domain to the end of the domain which you input ted before and parse it again. primary dns input your primary dns server address. alter dns input your standby dns server address. s elect it to use dhcp mode to get dns a ddress . i f you disable it, you will use static dns server. t he default is enabl ing it. if you uses pppoe mode you need to make the above setting. pppoe server it will be provided by isp. username input your adsl account. password input your adsl pa ssword. 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 lans networks id(for example, system will change la n 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 free datasheet http://
17 status and network id is also same as lans, phone will refuse to accept the ip to configure wan. 4.3.2. 2 . q o s confi g 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. t he vlan application of this phone is very flexible. in chart 1, there is a layer 2 switche s 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 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 i n blue vlan. b y this means, vlan divide the broadcast domain via restricting th e range of broadcast frame transmit ion . note: chart 2 use red and blue to identify the different vlan, but in practice, vlan uses different vlan ids to identify. free datasheet http://
18 qos configu ration field name explanation vlan enable before select it to enable vlan, you need enable bridge mode in l an config. vlan id check enable enable vlan id check by selecting it. after enable vlan id check, if vlan id of a data package is not the same wi th the phones or a data package do not have vlan id, the data 1 when others' vlan id doesnt match free datasheet http://
19 packets will discard. c ontrarily, the phone will accept the packets with the distinct vlan id. 7) you must gai n 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. service port field name explanation http port set web browse port, the default is 80 port quantity of rtp port, the default is 200. notice: 1 3 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 . free datasheet http://
20 sntp field name explanation server set sn tp server ip address. time zone select the time zone according to your location. time out set the time out, the default is 60 seconds. 12 hours systems switch the time mechanism between 12 hours and 24 hours. default is 24 hours mode sntp select the sntp, and click apply to make the sntp times effective. enable daylight enable daylight saving time time shift(minutes) setup the variety length month setup stat and end month week setup start and end week day setup start and end day hour setup s tart 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. free datasheet http://
21 sip config field name explanation choose line to set info a bout sip, there are 2 lines to choose. y ou can switch by load button . register status shows if the phone has been registered the sip server or not ; or so, show unapplied; server name set the server name. server address input your sip server address. server port set your sip server port. account name input your sip register account name. password input your sip register password. free datasheet http://
22 phone number input the phone number assigned by your voip service provider. phone will not register if there is no phone number configured. display name set the display name. proxy server address set proxy server ip address usually, register sip server configuration is the same as proxy sip server. but if your voip service provider give different configurations between register sip server and proxy sip server, you need make different settings. pr oxy server port set your proxy sip server port. proxy username input your proxy sip server account. proxy password input your proxy sip server password. domain realm set the sip domain if needed, otherwise this voip phone will use the register server a ddress 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. register expire time set expire time of sip server register, default is 60 sec onds. 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 recommended by the server, and register again . nat keep alive interval set examining interval of t he server, default is 60 seconds user agent set the user agent if have, the default is voip phone 1.0 signal key set the key for signal encryption media key set the key for rtp encryption local port s et sip port of each line r ing type s et ring type of each line subscribe expire time set the interval of subscribe. conference number set the server conference number to join the room enable dns srv s upport dns looking up with _sip.udp mode enable subscribe enable subscribe. enable keep authentication enable/disable keep authentication . nat keep alive enable/disable keep s 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 fun ction on. it need set the keep alive interval time less than the nat servers. enable via rport enable/disable system to support rfc3581. via rport is special way to realize sip nat. enable prack enable or disable sip prack function, suggest use the defa ult config. long contact s et more parameters in contact field; connection with sem 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; forward type 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. forward phone number appoint your forward phone number. free datasheet http://
23 server type select the special type of server which is encrypted, or has some unique requirements or call flows. dtmf mode select dtmf sending mode, there are three modes: ? dtmf_relay ? dtmf_rfc2833 ? dtmf_sip_info different voip service providers may provide different modes. rfc protocol edition select si p 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 change to rfc2543; else phone may not cancel call normally. system uses rfc3261 as default. transport protocol s et transport protocols , tcp or udp; rfc privacy edition set anonymous call out safely; support rfc3323and rfc3325 ; transfer expire time t he phone send bye and end the call as soon as hang up. enable conference number enable/disable conference enable d isplay name quote s et to make quotation mark to display name 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 enco de 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 jus t one codec which phone supports. auto tcp s et to use automatically tcp protocol to guarantee usability of transport as message is above 1300 byte enable strict proxy s upport 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 s et to support gruu 4.3.3. 2 . stun config in this web page, you can config sip stun. stun: by stun server, t he phone in private network could know the type of nat and the nat ma pping 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. free datasheet http://
24 stun field name explanation stun nat transverse shows stun nat transverse estima tion, true means stun can penetrate nat, while false means not. stun server addr set your sip stun server ip address stun server port set your sip stun server port stun effect time set stun effective time. if nat server finds that a nat 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. choose line to set info about sip, t here are 2 lines to choose. y ou can switch by load button . use stun 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 penetrati on 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 1 instead of 010 after we make a s etting 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. to save the memory and avoid abundant input of user, add the follow function s : free datasheet http://
25 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. i t m ay 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. dial peer field name explanation phone number there are two types of matching conditions: one is full matching, the other is prefix matching. in the full matching, you ne ed input your desired phone number in 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 numbe r to realize calling to what your prefix number is mapped. the prefix number supports at most 30 digits destination set destination address. this is optional config item. if you want to 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. port set the signal port, the default is 5060 for sip. alias set alias. this is optional config item. if you dont set alias, it will show no alias. free datasheet http://
26 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 t he 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. cal l mode select different signal protocol , sip suffix set suffix, this is optional config item. it will show no suffix if you dont set it. delete length set delete length. this is optional config item. for example: if the delete length is 3, the phone wil l 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 9t mapping: if yo u have registered a sip1 server and set dial - pee r 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, destination, alias and delete l ength. phone number is xxxt; destination is 255.255.255.255 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 d elete length. if you dial 93333, the sip2 server will receive 3333 this setting will realize speed dial function, after you dialing the numeric key 2, the number after all will be sent out. when you dial 2, the sip1 server will receive 3333444 4 free datasheet http://
27 the phone will automatically send out alias number adding your dialed number, if your dialed number starts with your set phone number. when you dial 8309, the sip1 server will receive 07558309 you need set phone number, alias and delete len gth. pho ne number is xxxt and alias is r ep: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. when you di al 0106228, the sip1 server will receive 86106228 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 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. free datasheet http://
28 dsp configuration field name explanation first codec the fist preferential dsp codec: g.711a/u, g.722, g.723, g.729, g.726 second codec the s econd preferential dsp codec: g.711a/u, g.722, g.723, g.729,g.726 third codec the third preferential dsp codec: g.711a/u, g.722, g.723, g.729,g.726 forth codec the forth preferential dsp codec: g.711a/u, g.722, g.723, g.729,g.726 fifth codec the fifth p referential dsp codec: g.711a/u, g.722, g.723, g.729, g.726 sixth codec 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 g729 payload length set g729 payload length handdown time specify the least reflection time of handdown, the default is 200ms. output volume specify output (receiver) volume grade. ring volume specify ri ng volume grade g722 timestamps 160/20ms or 320/20ms is available g723 bit rate 5.3kb/s or 6.3kb/s is available default ring type s et up the ring by default signal standard select signal standard. vad select it or not to enable or disable vad. if e nable 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 o n. free datasheet http://
29 call service field name explanation hotline specify hotline number. if you set the number, you can not dial any other numbers. no answer time specify no answer time p2p ip prefix 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 to reach 192.168.1.119. default is .. i f there is no . set, it means to disable dialing ip. enable call transfer enable call transfer by selecting it. enable call w aiting enable call waiting by selecting it. enable three way call enable three way call accept any call if select it, the phone will accept the call even if the called number is not belong to the phone. auto answer if select it, the phone will auto answ er when there is a n incoming call. ban outgoing if you select ban outgoing to enable it, and you can not dial out any number. auto handdown the phone will hang up and return to standby automatically at hands - free mode a uto handdown time a fter this tim e, the phone will hang up and return to standby automatically at hands - free mode do no t 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. black list 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 w ith 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 inco ming, he may add the number(s) to the list as the white list rule. the configuration rule is - number, for example, - 123456, or - 1234xx means any incoming number is forbidden except for 4119 note: end with dot (.) when set up the white l ist limit 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 l ist, 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 ar bitrary 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. free datasheet http://
30 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). use r 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. i n 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. digital map configuration field name explanation end with "#" set enable/disable the phone ended with # dial. free datasheet http://
31 911: cause 911 to be dialed immediately after it is entered. 99t4: cause 9 9 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 numb er according to your set rules. 4.3.4. 4 . function key configuration this phone supports 10 memory keys for speed dial. y ou could save 10 numbers from f1 to f10. t hen you could lift handset and press fn number to dial the number directly. 4.3.5. maint enance 4.3.5.1. auto provision auto provision field name explanation current config version show the current config files 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 free datasheet http://
32 keep blank. password set ftp server password. config file name set configuration files name which need to update. system will use mac as config file name if config fi le 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. update mode different update modes: 1. disable: means no update 2. update after reboot: means update after reboot. 3. update at time interval: means periodic update. enable dhcp option 66 if t his option is enabled, tftp server address defaults to the value of option 66 4.3.5.2. syslog config syslog i s 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 administr ator 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 sy stem 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. leve l 5 --- notice: your system works well in special condition, but you need to check its wo rking environment and parameter. level 6 --- info: the daily debugging info. level 7 --- debug: the lowest debug info. professional debugging info for r&d person. at present, the lowest level of debug information send to syslog is info, debug level only can b e displayed on telnet. syslog configuration field name explanation server ip set syslog server ip address. server port set syslog server port. mgr log level set the level of mgr log. sip log level set the level of sip log. enable syslog select it or not to enable or disable syslog. free datasheet http://
33 4.3.5.3. config setting config setting field name explanation save config you can save all changes of configurations. click the save button, all changes of configuration will be saved, and be effective immediat ely. . backup config right clicks on right click here and select save target as. then you will save the config file in .txt format clear config user can restore factory default configuration and reboot the phone. if 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 for voip accounts (sip1 - 2 ) and version number. 4.3.5.4. update you can update your configuration with your config file in thi s web page. update field name explanation web update click the browse button, find out the config file saved before or provided by manufacturer, download it to the phone directly, press update to save. you can also update downloaded update file, free datasheet http://
34 can be ip address or domain name with subdirectory. username set the ftp server username for download/upload. password set the ftp server password for downl oad/upload. file name set the name of update file or config file. the default 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. type 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 sel ect ftp/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 account configuration field name explanation keyboard password set the password for entering the setting m enu of the phone by the phones key board. the password is digit. this table shows the current user existed. user name set account user name. user level set user level, root user has the right to modify configuration , free datasheet http://
35 general can only read. password set the password. confirm 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 if you modified some configurations which need the phones 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 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. field name explanation mmi fileter ip table list: add or delete the ip address segments that acc ess 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. free datasheet http://
36 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 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 type s 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. firewal l, 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. field name explanation in_access enable select it to enable in_ access rule out_access enable select it to enable out_ access rule input/output specify current adding rule by selecting input rule or output r ule. free datasheet http://
37 deny/permit specify current adding rule by selecting deny rule or permit rule. protocol type filter protocol type. you can select tcp, udp, icmp, or ip. port range set the filter port range src addr set source address. it can be single ip address, network address, complete address 0.0.0.0, or network address similar to *.*.*.0 des addr set the destination address. it can be ip address, network address, complete address 0.0.0.0, or network address similar to *.*.*.* src mask set the source addres s 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. des mask set the destination address mask. for example, 255.255.255.255 means just point to one host; 255.255.255.0 me ans point to a network which network id is c type. click the add button if you want to add a new output rule. 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 re quest 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. click the delete button to delete the selected rule. 4.3.7. logout click logout and you will exit web page. if y ou want to enter it next time, you need input user name and password again. free datasheet http://
5 . appendix 5 .1. specification 5 .1.1. device specification item this voip phone ad a pter(input/output) input : 100 - 240vac 50 60hz output :5v/1 a port wan 10/100base - t rj - 45 for lan , auto mdix lan 10/100base - t rj - 45 for pc , auto mdix power consumption idle : 1. 5 w/active : 1.8w operation temperature 0 40 relative humidity 10 65% main chipset broadcom sdram 8 mbits flash 2 mbits size w x h x d 11.6 83 in.(29520575m m) weight 2.07lb.(0.94kg) 5 .1.2. voice features ? support 2 lines sip, sip 2.0 (rfc3261) ? codec g.711a/u g.7231 high/low g.729 , g.722,g.726 ? echo cancellation support g.168 and hand - free can support 96ms ? support vad cng ? nat transverse: support stun ? suppor ts full duplex. ? sip support sip domain sip authentication none basic md5 dns name of server, peer to peer ? sip support 2 server s , user can through each server to calling in and out ? dtmf : sip info dtmf relay rfc2833 ? sip application: contain sip call forwar d/transfer/holding/waiting / 3 way conference/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 distu rb and so on. ? s upport path, gruu ? support sip privacy. 5 .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: contai n ping t race route telnet client 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 ? sup port auto provisioning ? can config through web, telnet ? can upgrade firmware and configuration file through http, ftp, tftp ? support syslog free datasheet http://

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