EP 201 IP Phone User manual VoIP Phone EP 201 User Manual
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. z 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. z Before using the external power supply in the package, please check with home power voltage. Inaccurate power voltage may cause fire and damage.
Table of Content 1. INTRODUCING VI2006 VOIP PHONE.............................................................................................. 5 1.1. THANK YOU FOR YOUR PURCHASING EP201 ...................................................................................... 5 1.2. DELIVERY CONTENT .......................................................................................................................... 5 1.3. KEYPAD.............................................................................
.3.4.1. DSP Config .................................................................................................................... 33 4.3.4.2. Call Service.................................................................................................................... 34 4.3.4.3. Digital Map Configuration ............................................................................................ 36 4.3.4.4. Phone Book ................................................................................
1. Introducing IP 201 VoIP Phone 1.1. Thank you for your purchasing IP 201 Thank you for your purchasing EP 201, It’s 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 it own many data services features which you could not expect from a traditional telephone.
The power supply The Ethernet cable 1.3. Keypad The numeric keypad with the keys 0 to 9, *, and # is used to enter Digits and letters, additionally, the following keys are available: Key mapping: Key Key name Menu Phone Book Callers LED System Information Confirm Exit Navigation Key MWI Function Description In idle state press the MENU key to call up the menu. In idle mode, press the Phone Book key to check the record list and add new records and revise the record.
Use the key to realize blind transfer or attended transfer please refer to 3.1.4.-call transfer for more details). Transfer Conference Delete Hold Mute Use this key to realize the three party call ( pleae refer to3.1.5-Calling Hold and 3 ways call for more details) In menu, use this key to modify current setting or delete invalid information.
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.
Step 2: Connect the handset to the handset port by the handset cable in the package. Step 3: connect the power supply plug to the DC port on the back of the phone. Use the power cable to connect the power supply to a standard power outlet in your workspace. Step 4: push the on/off switch on the back of the phone to the on side, then the phone’s LCD screen displays “WAIT LOGON”. Later, a ready screen typically displays the date, time and current network mode.
key, the LCD screen will display “STATIC NET”. Then press the it by the key again, enter key, the LCD screen will display “USER NAME”. 5. Press the press the number. key and then press the key, input your PPPOE account number then key to confirm. The LCD screen will display the inputted PPPOE account key to return to the previous menu, then press the key, the LCD screen 6. Press the will display “PASSWORD”.
the key, input your desired IP address for your IP phone and confirmed by pressing the key, then the LCD will display the inputted IP address. When inputting IP with keypad, use “*” instead of “.”. key to return to previous menu, then press the 6. Press the display “DNS2”. Press the confirm it by pressing the 7. Press the key, input your spare DNS address and key, and then the LCD will display the inputted DNS address. key to return to previous menu, then press the display “DNS”.
Press the numeric key 2 and hold till the LCD screen displays “ARE YOU SURE”. Press the key, the LCD screen will display “CHANGING” and this VoIP phone is trying to switch to DHCP mode. If the icom “DHCP” on the top of the screen keeps blink, it shows that the phone is trying to access key to display the current IP; if the icon the DHCP server., and the IP is 0.0.0.0 if you press “DHCP” is showed without blink, it means that the phone has already gotten IP from DHCP server. 3. Basic Functions 3.1.
and use hands-free automatically. z Use handset Pick up the handset, and the LCD screen will display “PLEASE DIAL” and you will hear dialing tone at the same time, then input the phone number and end by the # button. When you hear long ring “du, du…” from handset and the LCD screen display “CALLING”, the call is through. Hang up the handset to end the call.
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 presses the key to hold A, and talk to C. Then B key, A will get through to C. 3. When A talks to B, B presses the A will get through to C. key, dial C phone number and # key, then hang up and 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.
the other missed calls or you can press the key again, the LCD screen will display the time of the missed calls. If there is no one missed calls, the LCD will display “LIST IS EMPTY”. z Received Calls key, and then the / key, till the LCD screen display “RECEIVED”. Press the Press ENTER key, the LCD screen will display the received call number and sequence number of the received call.
*1* means appointed prefix code. After making the above configuration, C can dial *1* plus B’phone number to pick up A’s call. User can set prefix in random, in the case of no affecting current dialing rules. 3.2.3. join call When B is calling C, A can join in the existing call by inputing an appointed prefix numbers plus B or C number, if B or C also supports join call The following chart shows how to configure an appointed prefix in dialpeer to have join call function. *2* means appointed prefix code.
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 phone key. The initial password is 123 for setting via phone key. The web browser on PC Telnet This Manual will tell you about the setting methods via the web browser on PC. 4.2.
Status Field name Network Phone Number Explanation Shows the configuration information on WAN and LAN port, including the connect mode of WAN port (Static, DHCP, PPPoE), MAC address, the IP address of WAN port and LAN port, ON or OFF of DHCP mode of LAN port. Shows the phone numbers provided by the SIP LINE 1-2 servers. The last line shows the version number and issued date. 4.3.1.2. Wizard Wizard Field Name Explanation Please select the proper network mode according to the network condition.
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 Address Netmask Gateway DNS Domain Primary DNS Alter DNS Display Name Server Address Server Port User Name Password Phone Number Enable Register Input the IP address distributed to you. Input the Netmask distributed to you. Input the Gateway address distributed to you. Set DNS domain postfix.
PPPoE Server It will be provided by ISP. Username Input your ADSL account. Password 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. Call Log Field name Start Time Last Time Called Number explanation Display the start time of the outgoing call Display the conversation time of the outgoing call.
WAN Config Field Name Active IP Current Netmask MAC Address Current Gateway Get MAC Time explanation The current IP address of the phone. The current Netmask address. The current MAC address of the phone. The current Gateway IP address. Shows the time of getting MAC address Please select the proper network mode according to the network condition.
DNS Domain Primary DNS Alter DNS 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. Input your primary DNS server address. Input your standby DNS server address. 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 password.
In chart 1, there is a layer 2 switch without setting VLAN. Any broadcast frame will be transmitted to the other ports except the send port. For example, a broadcast information is sent out from port 1 then transmitted to port 2,3and 4. In chart 2, red and blue indicate two different VLANs in the switch, and port 1 and port 2 belong to red VLAN, port 3 and port 4 belong to blue VLAN.
Voice/Data VLAN differentiated DiffServ Enable DiffServ Value Voice 802.1P Priority Data 802.1P Priority Voice VLAN ID Data VLAN ID ID; tag differentiated means after using VLAN, VoIP(signal and voice) packets will add voice VLAN ID, and other data packets will add data VLAN ID; data untaged means after using VLAN, only VoIP packets will add voice VLAN ID. Other data packets will not use VLAN. Select it or not to Enable or disable DiffServ. Set DiffServ value, the common value is 0x00. Specify 802.
HTTP Port Telnet Port enhance system safety you'd better change it into non-80 standard port Example: The IP address is 192.168.1.70. and the port value is 8090, the accessing address is http://192.168.1.70:8090 Set Telnet Port, the default is 23. You can change the value into others. Example: The IP address is 192.168.1.70. the telnet port value is 8023, the accessing address is telnet 192.168.1.70 8023 Set the RTP Initial Port. It is dynamic allocation.
Time shift(minutes) Month Week Day Hour Minute Setup the variety length Setup stat and end month Setup start and end week Setup start and end day Setup start and end hours 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.
SIP Config Field name explanation 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; 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.
Register Expire Time NAT Keep Alive Interval User Agent Signal Key Media Key Local port Ring type Subscribe Expire Time Conference Number Enable DNS SRV Enable Subscribe Enable Keep Authentication NAT Keep Alive Enable Via rport Enable PRACK Long Contact Enable URI Convert Dial Without Register Ban Anonymous Call Forward Type Forward Phone Number Server Type DTMF Mode RFC Protocol Edition Transport Protocol RFC Privacy Edition Set expire time of SIP server register, default is 60 seconds.
Transfer Expire Time Enable Conference Number Enable Displayname Quote Click to Talk Signal Encode RTP Encode Enable Session Timer Answer With Single Codec Auto TCP Enable Strict Proxy Enable GRUU RFC3325; The phone send bye and end the call as soon as hang up. Enable/Disable conference Set to make quotation mark to displayname as the phone sends out signal, in order to be compatible with server. Set click to Talk ( need practical software support). Enable/Disable Signal Encrypt.
STUN Field name Explanation STUN NAT Transverse Shows STUN NAT Transverse estimation, true means STUN can penetrate NAT, while False means not. Set your SIP STUN Server IP address Set your SIP STUN Server Port 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. Set the SIP port.
To save the memory and avoid abundant input of user,add the follow fuctions: 1 x Match any single digit that is dialed. If user makes the above configuration, after user dials 11 digit numbers started with 13, the phone will send out 0 plus the dialed numbers automatically. 2 [] Specifies a range that will match digit. It may be a range, a list of ranges separated by commas, or a list of digits.
will show no alias. Note: There are four types of aliases. 1) add: xxx, it means that you need dial xxx in front of phone number, which will reduce dialing number length. 2) all: xxx, it means that xxx will replace some phone number. 3) del: It means that phone will delete the number with length appointed. 4) Rep: It means that phone will replace the number with length and number appointed.
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 33334444 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 Length.
DSP Configuration Field name explanation First Codec The fist preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726 The second preferential DSP codec: G.711A/u, G.722, G.723, G.729,G.726 The third preferential DSP codec: G.711A/u, G.722, G.723, G.729,G.726 The forth preferential DSP codec: G.711A/u, G.722, G.723, G.729,g.726 The fifth preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726 The sixth preferential DSP codec: G.711A/u, G.722, G.723, G.729, G.726 Specify Input (MIC) Volume grade.
Call Service Field name Explanation Hotline MWI Number Enable Call Transfer 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 to reach 192.168.1.119. Default is “.”. If there is no “.” Set, it means to disable dialing IP. Set the number to listen voice mail in server.
example, 6. expresses any number with prefix 6 will be forbidden to dialed 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 Means any incoming number is forbidden except for 4119 Note: End with DOT (.) when set up the white list Limit List Set Add/Delete Limit List. Please input the prefix of those phone numbers which you forbid the phone to dial out.
Digital Map Configuration Field name explanation End with "#" Fixed Length Set Enable/Disable the phone ended with “#” dial. Specify the Fixed Length of phone ending with. Set the timeout of the last dial digit. The call will be sent after timeout. Time out 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. .
Phone Book Field name Explanation Shows the detail of current phonebook. Shows the name corresponding to the phone number Name Number Shows the phone number Ring Type Shows the ring type of the incoming call. Click “Modify” to change the selected information and click the “Delete” to delete the selected record. Notice: the maximum capability of the phonebook is 500 items 4.3.5. Maintenance 4.3.5.1.
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.
Config Setting Field name Save Config Backup Config Clear Config explanation you can save all changes of configurations. Click the Save button, all changes of configuration will be saved, and be effective immediately. . Right clicks on “Right click here…” and select “Save Target As….” then you will save the config file in .txt format user can restore factory default configuration and reboot the phone.
address can be IP address or Domain name with subdirectory. Set the FTP server Username for download/upload. Set the FTP server password for download/upload. 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.
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 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..
segment. 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).
Select it to Enable out_ access rule Specify current adding rule by selecting input rule or output rule. Deny/Permit Specify current adding rule by selecting Deny rule or Permit rule. Filter protocol type. You can select TCP, UDP, ICMP, or IP. Protocol Type Port Range Set the filter Port range Set source address. It can be single IP address, network address, Src Addr complete address 0.0.0.0, or network address similar to *.*.*.0 Set the destination address.
--Config-Network --Config-System --Config-DSP 5. Appendix 5.1. Specification 5.1.1. Device specification Item Adapter(Input/Output) WAN Port LAN Power Consumption this VoIP Phone Input:100-240VAC 50 60Hz Output:5V/1A 10/100Base- T RJ-45 for LAN, Auto MDIX 10/100Base- T RJ-45 for PC, Auto MDIX Idle:1.5W/Active:1.
LCD size Operation Temperature Relative Humidity Main Chipset SDRAM Flash 74 x 28mm Size W x H x D Weight 11.6×8×3 in.(295×205×75mm) 2.07lb.(0.94kg) 0 40 10 65% Broadcom 8Mbits 2Mbits 5.1.2. Voice Features z z z z z z z z z z z z z z z z z z z z 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 Supports full duplex.
5.2. Digit-character map table Button Character 1@ Button Character 7PQRSpqrs 2ABCabc 8TUVtuv 3DEFdef 9WXYZwxyz 4GHIghi .
