Fanvil Product User Manual IP Phone Model: C56/C56P Version: V.2.2.0.0 © 2005 Fanvil technology Co., Ltd All rights reserved. This document is supplied by Fanvil Technology Co.
Table of Content 1.Introducing C56/C56P VoIP Phone.............................................................................................................4 1.1. Thank you for your purchasing C56/C56P .....................................................................................4 1.2. Delivery Content................................................................................................................................4 1.3. Keypad..........................................................
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. Before using the external power supply in the package, please check with home power voltage. Inaccurate power voltage may cause fire and damage.
1. Introducing C56/C56PVoIP Phone 1.1. Thank you for your purchasingC56/C56P Thank you for your purchasing C56/C56P, C56/C56P 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 enjoying other features that traditional phone has, but also it own many data services features which you could not expect from a traditional telephone.
Hold mute Redial Handfree Temporarily hold the active call during the talking; press the key again to unhold the call. You also can press this key then input the third party’s phone number and end with the # key during calling; you can make a call with the third party and hold the previous calling. (3.1.5-Calling Hold and 3 ways call).
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.
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.
10. Press the key twice, then press numeric key “3”and hold until the screen display “ARE YOU SURE”. Press the key, the screen will display “CHANGING”, which means that the phone is trying to switch to PPPoE mode.
display “DNS”. Press the pressing the 8. Press the key then the key, input your DNS address and confirm it by key, and then the LCD will display the inputted DNS address. key to return to the previous menu, and then press the will display “GATEWAY”. Press the confirm it by pressing the 9. Press the key and the key, the LCD screen key, input your gateway’s IP address and key, the LCD screen will display the inputted gateway address.
server, and the IP is 0.0.0.0 if you press key to display the current IP; if the icon “DHCP” is showed without blink, it means that the phone has already gotten IP from DHCP server. 3. Basic Functions 3.1. Basic operation 3.1.1. Accepting a call There are four methods to accept an incoming call: Pick up handset to accept incoming calls. Press the button If you need switch from a hands-free call to handset, please pick up the handset directly.
3.1.3. Ending a call Hangs up by handset on hook Hangs up by press when in hands-free Hangs up a call in call waiting state. If you are in call waiting state, you could press # key to hang up the current call, and switch to the other call to keep talking. Note: Pressing # key will not hang up if there is only one call currently. 3.1.4. Transferring a call Call transfer has several ways to realize: 1.
Missed Calls Press the the missed call. key, and then the key, till the LCD screen display “MISSED”. Press key, the LCD screen will display the missed call number and sequence numbers of the You can press the / key to dial this phone number, you also press UP/DOWN key to browse the other missed calls or you can press the key again, the LCD screen will display the time of the missed calls. If there is no one missed calls, the LCD will display “LIST IS EMPTY”.
3.2.2. redial/unredial If B is in busy line when A calls B, A will get notice: busy, please hang up. If A want to connect Bas soon as B is in idle, he can use redial function at the moment and he can dial 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.
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 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.
4.3. Configuration via WEB 4.3.1. BASIC 4.3.1.1. Status Status Field name Network Accounts 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 Field Name Explanation Please select the proper network mode according to the network condition. FV6030 provide three different network settings: Static IP: 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.
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. 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。 Server Names It will be provided by ISP. User Input your ADSL account. Password Input your ADSL password.
Field name Start Time Duration Dialed Calls explanation Display the start time of the outgoing call Display the conversation time of the outgoing call. Display the account/protocol/line of the outgoing call. 4.3.2. Network 4.3.2.1. WAN Config WAN Config Field Name Active IP Address Current Subnet Mask MAC Address Current IP Gateway MAC Timestamp explanation The current IP address of the phone. The Current Subnet Mask address. The current MAC address of the phone. The current Gateway IP address.
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 Address Input the IP address distributed to you. Subnet Mask Input the Subnet Mask distributed to you. IP Gateway Input the Gateway address distributed to you. Set DNS domain postfix.
In chart 1, there is 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 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.
Audio DSCP Enable WAN Port VLAN WAN Port VLAN ID SIP 802.1P Priority Audio 8021P Priority LAN Port VLAN Mode LAN Port VLAN ID Specify the value of the Audio DSCP Enable WAN Port VLAN by selecting it Specify the value of the WAN Port VLAN ID, the range of the value is 0-4095 Specify the value of the voice 8021.p priority, the range of the value is 0-7 Specify the value of the signal 8021.
Notice: 1）You need save the configuration and reboot the phone after set this page. 2）If you modify the port of Telnet and HTTP, you would better set the value more than 1024 because the port value less than 1024 is system port reserved. 3）if you set 0 for the HTTP port, it will disable HTTP service. 4.3.2.4. TIME&DATE 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.
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.
SIP Config Field name explanation Choose the sip line to set info about SIP;there are 2 lines to choose. You can switch by 【Load】 button. 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. Authentication User Input your SIP registered account name. AuthenticationPassword Input your SIP registered password.
Proxy Server Address Proxy Server Port Proxy User Proxy Password Domain Realm Backup Server Address Backup Server Port Enable Registeration Disable Codecs/Enable Codecs Forward Type Forward Number No Ans.
Subscribe Period Conference Type Conference Number Registration Expires Enable Service Code DND On Code DND Off Code Always CFwd On Code Always CFwd Off Code Busy CFwd On Code Busy CFwd Off Code No Ans. CFwd On Code No Ans. CFwd Off Code Anonymous On Code administrator for the connecting code. Different systems have different codes. Overtime of resending subscribe packet. Suggest using the default configuration.
Anonymous Off Code Keep Alive Type Keep Alive Interval User Agent DTMF Mode Local port Ring type Enable Rport Enable PRACK Long Long Contact Convert URI Dial Without Registered Ban Anonymous Call Enable DNS SRV Server Type RFC Protocol Edition Transport Protocol Anonymous Call Edition Keep Authentication Ans.
Click to Talk Enable BLF List BLF List Number Strict Branch Enable Group Registration Failure Retry Time Set click to Talk (need practical software support).
STUN Field name explanation STUN NAT Transversal Shows STUN NAT Transverse estimation, true means STUN can penetrate NAT, while False means not. Server Address Set your SIP STUN Server IP address Server Port Set your SIP STUN Server Port Set STUN blinding period(s). If NAT server finds that a NAT Binding Period 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.
To save the memory and avoid abundant input of user, add the follow functions: 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.
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.
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. AUDIO In this page, you can configure voice codec, input/output volume and so on.
4.3.4.2. FEATURE In this web page, you can configure Hotline, Call Transfer, Call Waiting, 3 Ways Call, Black List, white list Limit List and so on.
Call Service Field name Do Not Disturb Ban Outgoing Enable Call Transfer Semi-Attended Transfer Enable Call Waiting Enable 3-way Conference Accept Any Call Enable Auto Handdown Auto Handdown Time Enable Silent Mode Enable Intercom Enable Intercom Mute Enable Intercom Tone Enable Intercom Barge Turn Off Power Light DND Return Code Busy Return Code Reject Return Code P2P IP Prefix Active URI Limit IP Action URL Settings Block Out Settings explanation Select NO Disturb, the phone will reject any incoming
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.
Cause extensions 1000-8999 to be dialed immediately Cause 8 digit numbers started with 9 to be dialed immediately Cause 911 to be dialed immediately after it is entered. Cause 99 to be dialed after 4 seconds. 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.
added to the phone.
4.3.5. FUNCTION KEY The phone has 4 programmable keys which are able to set up to many functions per key. The following list shows the functions you can set on the programmable keys and provides a description for each function. The default configuration for each key is none which means the key hasn’t been set for any functions. 1. Set the type as Memory Key When the type is memory key, you can input number in value input, and phone will call the inputted number as pressing DSS key. 2.
4.3.6. Maintenance 4.3.5.1. Auto Provision Fanvil endpoint supports PnP and DHCP and Phone Flash to obtain the parameters. The PnP and DHCP and Phone Flash are all deployed, endpoint will go by the following process to try to obtain the server address and other parameters, when it boots up: DHCP option PnP server Phone Flash Auto Provision Field name Auto Provision Setting Current Config Version Common Config Version explanation Show the current config file’s version.
User Password Config Encrypt Key Common Config Encrypt Key DHCP Option Settings DHCP Option Setting Custom DHCP Option Plug and Play Settings Enable PnP PnP Server PnP Port PnP Transport PnP Interval Phone Flash Settings Server Address Protocol Type Config File Name Update Interval Update Mode provision would stop here.
Syslog Configuration Field name Syslog Settings Server IP Server Port MGR Log Level SIP Log Level IAX2 Log Level Enable Syslog Web Capture Start Stop explanation Set Syslog server IP address. Set Syslog server port. Set the level of MGR log. Set the level of SIP log. Set the level of IAX2 log. Select it or not to enable or disable syslog.
Config Setting Field name Save Configuration Backup Configuration Clear Configuration 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 config File(.txt)” then you will save the config file in .txt format, or select “Save Target As config File(.xml)” then you will save the config file in .
Protocol FTP/TFTP server. The configuration will be effective after the phone is reset. 4. Phone book export (.vcf, .csv, .xml): Upload the phonebook file to FTP/TFTP server, name and save it. 5. PhoneBook import (.vcf, .csv, .xml): Download the phonebook file to phone from FTP/TFTP server.
4.3.5.5. Access You can add or delete user account, and change the authority of each user account in this web page Access Configuration Field name LCD Menu Password Settings explanation Set the password for entering the setting menu of the phone by the phone’s key board. The password is digit. This table shows the current user existed. User Set account user name. User Level Set user level, Root user has the right to modify configuration, General can only read. Password Set the password.
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 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.
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 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.
which network ID is C type. Set the destination address’ mask. For example, 255.255.255.255 Dest 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. 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.
4.4.2. Phone menu Phone main menu: 5. Appendix 5.1. Specification 5.1.1. Device specification Item Adapter(Input/Output) Port this VoIP Phone Input：100-240VAC 50～60Hz Output：5V/1A WAN 10/100Base- T RJ-45 for LAN, Auto MDIX LAN 10/100Base- T RJ-45 for PC, Auto MDIX Power Consumption LCD size Idle：1.5W/Active：1.8W 74 x 28mm Operation Temperature 0～40℃ Relative Humidity 10～65% Main Chipset broadcom voip chipset SDRAM 8MB Flash 2MB Size（W x H x D） Weight 20（18.5）x19.3cm 0.99kg 5.1.2.
Support phonebook 500 records, incoming calls / outgoing calls / missing calls. Each supports 100 records support conference call in server Could dial use private server automatically when public server unregistered while private server is registered successfully Phonebook supports VCard standard Support 12/24 time format. 12/24 hours time display Support daylight saving time Support path, gruu Support SIP Privacy. 5.1.3.
4GHIghi .
