Fanvil C56 User Manual

Fanvil C56 User Manual

Fanvil c56/c56p ip phone user manual
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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., Ltd, No part of this document may
be reproduced, republished or retransmitted in any form or by any means whatsoever,
whether electronically or mechanically, including, but not limited to, by way of
photocopying, recording, information recording or through retrieval systems, without
the express written permission of Fanvil Technology Co., Ltd. Fanvil Technology Co., Ltd
reserves the right to revise this document and make changes at any time and without
the obligation to notify any person and/or entity of such revisions and/or changes.
Product specifications contained in this document are subject to change without notice.

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Summary of Contents for Fanvil C56

  • Page 1 © 2005 Fanvil technology Co., Ltd All rights reserved. This document is supplied by Fanvil Technology Co., Ltd, No part of this document may be reproduced, republished or retransmitted in any form or by any means whatsoever, whether electronically or mechanically, including, but not limited to, by way of photocopying, recording, information recording or through retrieval systems, without the express written permission of Fanvil Technology Co., Ltd.
  • Page 2: Table Of Contents

    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..............................4 1.4. Port for connecting..........................5 2. Initial connecting and Setting........................5 2.1. Connect the phone ..........................5 2.2. Initial Setting .............................7 2.2.1. PPPoE mode..........................7 2.2.2.
  • Page 3: Safety Notices

    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.
  • Page 4: Introducing C56/C56Pvoip Phone

    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.
  • Page 5: Port For Connecting

    Temporarily hold the active call during the talking; press Hold 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.
  • Page 6 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”.
  • Page 7: Initial Setting

    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.
  • Page 8: Static Ip Mode

    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. If the icon “PPPoE” on the top of the screen keeps blink, it shows that the phone is trying to access the PPPoE server, and the IP is still static IP if you press key to display the current IP;...
  • Page 9: Dhcp Mode

    display “DNS”. Press the key then the key, input your DNS address and confirm it by pressing the key, and then the LCD will display the inputted DNS address. 8. Press the key to return to the previous menu, and then press the key, the LCD screen will display “GATEWAY”.
  • Page 10: Basic Functions

    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.
  • Page 11: Ending A Call

    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.
  • Page 12: The High-Level Operation

     Missed Calls Press the 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 missed call. 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.
  • Page 13: Redial/Unredial

    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.
  • Page 14: Setting

    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.
  • Page 15: Configuration Via Web

    4.3. Configuration via WEB 4.3.1. BASIC 4.3.1.1. Status Status Field name Explanation Shows the configuration information on WAN and LAN port, Network including the connect mode of WAN port (Static, DHCP, PPPoE), MAC address, the IP address of WAN port and LAN port, ON or OFF of DHCP mode of LAN port.
  • Page 16 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.
  • Page 17 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.
  • Page 18: Network

    Field name explanation Start Time Display the start time of the outgoing call Duration Display the conversation time of the outgoing call. Dialed Calls Display the account/protocol/line of the outgoing call. 4.3.2. Network 4.3.2.1. WAN Config WAN Config Field Name explanation Active IP Address The current IP address of the phone.
  • Page 19  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.
  • Page 20: Qos Configuration

    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.
  • Page 21: Service Port

    Audio DSCP Specify the value of the Audio DSCP Enable WAN Port Enable WAN Port VLAN by selecting it VLAN WAN Port VLAN ID Specify the value of the WAN Port VLAN ID, the range of the value is 0-4095 SIP 802.1P Priority Specify the value of the voice 8021.p priority, the range of the value is 0-7...
  • Page 22 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.
  • Page 23: Voip

    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.
  • Page 24: Sip Config

    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.
  • Page 25 Set proxy server IP address(Usually, Register SIP Server configuration is the same as Proxy SIP Server. But if your VoIP Proxy Server Address service provider give different configurations between Register SIP Server and Proxy SIP Server, you need make different settings.)...
  • Page 26 administrator for the connecting code. Different systems have different codes. Subscribe Period Overtime of resending subscribe packet. Suggest using the default configuration. Conference Type Specify the Conference Type, if you select the local, you needn’t input the conference number Conference Number Specify the network conference number, please contact your system administrator for the network conference number Set expire time of SIP server register, default is 60 seconds.
  • Page 27 Anonymous Off Code Set the Anonymous Off Code, When you choose to disable the anonymous call function on your IP phone, it will send information to the server, and the server will disable the anonymous call function for your IP phone automatically. Keep Alive Type Specify the keep alive type, if the type is option, the phone will send option sip message to server every NAT Keep...
  • Page 28 Click to Talk Set click to Talk (need practical software support). Enable BLF List Enable BLF List by selecting it, BLF list is a function which can monitor the group status, it is not one to one monitoring, but the information feedback from the sever to decide which BLF list will monitor BLF List Number...
  • Page 29 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...
  • Page 30: Dial Peer

    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、...
  • Page 31 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.
  • Page 32: Phone

    If your dialed phone number When you dial “147”, starts with your set phone the SIP1 server will number. The phone will send receive “1470011” out your dialed phone number adding suffix number. 4.3.4. Phone 4.3.4.1. AUDIO In this page, you can configure voice codec, input/output volume and so on. DSP Configuration Field name explanation...
  • Page 33 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...
  • Page 34: Call Service

    Call Service Field name explanation Do Not Disturb Select NO Disturb, the phone will reject any incoming call, the callers will be reminded by busy, but any outgoing call from the phone will work well. Ban Outgoing If you select Ban Outgoing to enable it, and you cannot dial out any number. Enable Call Transfer Enable Call Transfer by selecting it.
  • Page 35: Digital Map Configuration

    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.
  • Page 36: Phone Book

    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.
  • Page 37 added to the phone. Export Contact File Export XML Click export xml button to export phonebook file of xml model Export CSV Click export xml button to export phonebook file of csv model Export VCF Click export xml button to export phonebook file of vcf model Blacklist Settings Type Select the blacklist type, you can select number or prefix of number...
  • Page 38: Function Key

    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.
  • Page 39: 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...
  • Page 40 provision would stop here. If the endpoints confirm the configuration by Digest method, the endpoints wouldn’t upgrade configuration unless the configuration in the server is different with the running configuration. User Set FTP/HTTP/HTTPS server Username. System will use anonymous if username keep blank. Password Set FTP/HTTP/HTTPS server Password.
  • Page 41: Syslog Configuration

    Syslog Configuration Field name explanation Syslog Settings 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. IAX2 Log Level Set the level of IAX2 log.
  • Page 42: Config Setting

    Config Setting Field name explanation Save Configuration You can save all changes of configurations. Click the Save button, all changes of configuration will be saved, and be effective immediately. Backup Configuration Right clicks on “Right click here…” and select “Save Target As config File(.txt)”...
  • Page 43 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. Protocol Select FTP/TFTP server...
  • Page 44: Access Configuration

    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 explanation LCD Menu Password Set the password for entering the setting menu of the phone by the Settings phone’s key board.
  • Page 45: Security

    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...
  • Page 46: Firewall Configuration

    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.
  • Page 47: Logout

    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.
  • Page 48: Appendix

    4.4.2. Phone menu Phone main menu: 5. Appendix 5.1. Specification 5.1.1. Device specification Item this VoIP Phone Adapter(Input/Output) Input:100-240VAC 50~60Hz Output:5V/1A 10/100Base- T RJ-45 for LAN, Auto MDIX Port 10/100Base- T RJ-45 for PC, Auto MDIX Power Consumption Idle:1.5W/Active:1.8W LCD size 74 x 28mm Operation Temperature 0~40℃...
  • Page 49: Digit-Character Map Table

     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 ...
  • Page 50 4 G H I g h i 5 J K L j k l 6 M N O m n...

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