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Fanvil C600

314.54
(374.30 inc tax)
10 days
FANVILC600
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The C600 provides the complete desk phone solutions with

flexible functionality, HD Voice Quality, and highly easy

operation. It is absolutely an excellent Smart Video Phone for

small to large sized business, as well as mission-critical

Enterprise Business.

Highlights

√ Full Programmable DSS Key and Soft Function Key

√ DSS Can Support Upto 100 Stations

√ On Line Recording & Message Forwarding

√ Call Conference Enhancement

√ Flexible Dial Plan with Auto Provision

√ Support LDAP & XML Phone Book

√ Support WIFI / Bluetooth (Optional)

Display
 7” TFT 800x480 Capacitive Multi Touch Screen

Call Features
 Call Forwarding
 Call Transfer (blind/attended/alert)
 Call Holding
 Call Waiting
 Call Conference
 BLF List
 Capable of 10 way conversation
 Join Call
 Pickup
 Call Completion
 Hot desk function
 Soft DSS Key (Upto 100)
 Auto Redial / un-redial
 Support multi line and pre-dial
 Support messaging and MWI
 Voice recording during talking /local
 Flexible dial plan
 Hotline/Warm-line
 Call Rejection
 Black List
 Barring function for outgoing calls
 Do Not Disturb
 Auto Answer (Hands- free / Headset)
 Caller ID display
 CLIR (rejects anonymous calls)
 CLIP (to make an anonymous call)
 Dial without registration
 Supports Call Logs with Missed / Incoming / Outgoing calls
 Intercom/Intercom barge
 Password dial
 Headset ring
 Direct IP call without SIP proxy

Phone features
 Android 4.2 OS 
 Supports high quality video call
 Supports SIP 2.0 (RFC3261) and correlative RFCs
 Supports 6 SIP servers and Backup SIP proxy servers.
 Supports SIP UDP/TCP/TLS
 Supports HDMI 
 Supports USB Host
 Supports Acoustic echo cancellation (AEC) - 128ms max filter length in duplex speaker phone mode
 Supports PLC & adaptive jitter buffer 

Audio Features
 Wideband codec: G.722
 Narrowband codec: G.711(A/μ), G.723.1, G.729AB, iLBC, AMR
 DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO 
 Full-duplex hands-free speaker phone with AEC (Max filter length – 128ms)
 Voice Activity Detection (VAD)
 Comfort Noise Generation (CNG)
 Adaptive jitter buffer
 Packet loss concealment (PLC)
 Audio format: WAV/MP3/OGG
Video Features
 Video codec: H.264 / H.263
 Image codec: JPEG/PNG/BMP/GIF
 Video format: MP4/3GP/FLV
 Video call resolution: QCIF / CIF / VGA / 4CIF (1280x720P Optional)
 Bandwidth selection: 64kbps~4Mbps
 Frame rate selection: 10~30fps
 Picture-in-Picture (PIP) 
 Video from remote site can be displayed in full screen
 Supports up to 4 video display mode 

Network/ Security Features
 WAN/LAN: support bridge mode
 Supports PoE (802.3af)
 Supports main DNS and secondary DNS server
 Supports VLAN
 Supports SNTP Client
 Supports VPN L2TP / PPTP / IPSec
 Supports SIP SRTP, 
 Supports Web HTTP / HTTPS
 Supports QoS: 802.1p/q, DSCP
 Supports MD5 authentication
 Supports Web Filter
 DHCP/ static/ PPPoE
 STUN

Maintenance & Management
 Android 4.2
 Supports third party communication APP
 Supports Web, Telnet 
 Web Management with different account right
 Supports automatic upgrades/ configuration deployment
 Supports encrypted configuration files download with AES
 Supports pushing message
 Supports Auto-Provisioning (DHCP option/ PnP/ Phone flash)
 Supports TR-069
 Supports Web upgrade
 Supports U Disk/SD upgrade
 Supports backup/restore/factory reset data
Physical Features
 Adapter Input:100-240V 
 Adapter Output:12V/1A
 WAN Port -10/100/1000 Base-T RJ-45 for LAN
 LAN Port- 10/100/1000 Base-T RJ-45 for PC
 HDMI Port - Type A
 SD Interface - TF Card Support Upto 32G
 USB Port- USB 2.0
 Power Consumption- Idle: 2.5W/Active: 5W
 LCD Size Diagonal:7 inch (800 x 480) Capacitive touch screen
 Camera- Adjustable Position
 Operation Temperature: 0~40℃
 Relative Humidity: 10~65%
 CPU Freescale Core Quad 1 Ghz
 SDRAM - 1GB DDR3 1066
 Flash- 4GB
 Weight- Phone:1.2Kg/Total:1.8Kg
 Call Forwarding
 Call Transfer(blind/attended/alert)
 Call Holding
 Call Waiting
 Call Conference
 BLF List
 Capable of 10 way conversation
 Join Call
 Pickup
 Call Completion
 Hot desk function
 Soft DSS Key (Upto 100)
 Auto Redial / un-redial
 Support multi line and pre-dial
 Support messaging and MWI
 Voice recording during talking /local
 Flexible dial plan
 Hotline/Warm-line
 Call Rejection
 Black List
 Barring function for outgoing calls
 Do Not Disturb
 Auto Answer (Hands- free / Headset)
 Caller ID display
 CLIR (rejects anonymous calls)
 CLIP (to make an anonymous call)
 Dial without registration
 Supports Call Logs with Missed / Incoming / Outgoing calls
 Intercom/Intercom barge
 Password dial
 Headset ring
 Direct IP call without SIP proxy

System Requirements

A Windows desktop or Windows server OS: Windows 7/8, Windows 10, Windows Server 2008 R2 with SP1, 2012 R2, 2016 R2) IPv4/IPv6 Windows firewall

This document assumes that the Windows OS is already deployed and administrators of PortSIP PBX are allocated the administrator permission to Windows.

 

Hardware and Software Dependencies

OS Supported by PortSIP PBX includes:

Linux Server:

CentOS 7 or higher, 64bit; gcc/g++ 6.4 or higher

Ubuntu 16.04.4 or higher, 64bit; gcc/g++ 6.4 or higher

Debian 9.0 or higher, 64bit; gcc/g++ 6.4 or higher

Windows Desktop:

Windows 7, 8 and 10, 64-bit

Windows Server:

Windows 2008 R2 SP1, 2012 R2, 2016 R2, 64-bit

Important: The OS must be up to date.

 

 

Cloud and Virtualization Environment Supported

To build high-availability communication solution to help clients reduce cost and improve communication performance, PortSIP PBX commits support on cloud services and have confirmed availability on following cloud and virtualization environment:

  • VMware ESX 5.X and above.
  • Linux HyperV
  • Microsoft HyperV 2008 R2 and above
  • Amazon AWS
  • UCloud
  • Alibaba Cloud
  • Linode
  • Digital Ocean
  • Godaddy VPS and Cloud
  • Tencent Cloud

 

System performance depends on following key factors:

  • Maximum simultaneous calls needed for PBX
  • Maximum online users needed for PBX
  • Recordings for calls
  • Record audio only or both of audio, video
  • Maximum online users for audio/video conferences on PBX
  • Maximum IVR (Virtual Receptionist) on PBX
  • Maximum Call Queues on PBX
  • Maximum Ring Groups on PBX

Depending on the key features listed above, PortSIP PBX is able to run on PCs and servers with various CPSs ranging from Intel i3 CPU to Inter Xeon

 

Other Requirements

  • Latest Firefox, Google Chrome or Internet Explorer
  • Microsoft .NET Framework version 4.5 or higher
  • Knowledge of Linux and Linux Internet administration
  • Knowledge of Windows and Windows Internet administration
  • A constant internet connection to service.portsip.com on port 6881.
  • A constant internet connection to stun.portsip.com and stun1.portsip.com on port 3478.
  • A constant internet connection to stun4.l.google.com on port 3478.

PortSIP PBX Features

PortSIP PBX free Edition and Full Edition are equipped with same features, with the only difference that the free Edition only support up to 3 simultaneous calls only.

 

Features Free Edition Full Edition
Extensions Unlimited Unlimited
Number of Simultaneous Calls Supported 3 >10,000
Linux Support (CentOS, RHEL, Debian, Ubuntu) Y Y
Multi-Tenant Y Y
Call Logging Y Y
Call Forward on Busy or No Answer Y Y
Call Routing by DID Y Y
Auto Attendant / Digital Receptionist Y Y
Voicemail/ Music on Hold Y Y
Central Phonebook TBD TBD
Call Transfer Y Y
MWI – Message Waiting Indicator Y Y
Ring Extension & Mobile Simultaneously Y Y
Automatic Pickup on Busy Y Y
Supports SIP Trunks/ Gateways Y Y
Custom SMTP Server Y Y
Custom FQDN Y Y
Busy Lamp Field (BLF) Y Y
Call Reporting Y Y
Call Parking / Pickup Y Y
Call Queuing Y Y
Audio Call Recording Y Y
Video Call Recording Y Y
Intercom/ Paging Y Y
Call Recordings Management Y Y
Configure BLF’s from the Clients Y Y
Web-based Management Console Y Y
Automated Provisioning of Devices Y Y
Real Time Web-based System Status Y Y
Integrated Web Server Y Y
Easy Backup and Restore Y Y
VMware / Hyper-V Compatibility Y Y
Scheduled Backup Y Y
Scheduled Restore Y Y
Inbuilt Fail Over Functionality Y Y
Standby License Y Y
Deploy as cluster Y Y
See the Presence of Your Colleagues Y Y
Receive Voice Mail via Email Y Y
Advanced Forwarding Rules Y Y
Setting Up Conference Calls Y Y
Click2Call Y Y
View Presence of Remote Offices Y Y
Advanced Queue Strategies Y Y
Advanced Call Reporting Y Y
Real Time Queue Statistics Y Y
Ability to Use PortSIP VoIP SDK Y Y
Real Time Queue Monitoring Y Y
Call Recordings Search Y Y
Supports External Agents Y Y
Android Client (Provide rebrand/OEM) Y Y
iOS Client (Provide rebrand/OEM) Y Y
Windows Client Y Y
Mac Client Y Y
Web client Y Y
Mobile PUSH Y Y
Avoid NAT Problems Y Y
Automatic Plug & Play Phone Provisioning Y Y
Manage IP Phones Network Wide from Console Y Y
Restart Phones Remotely Y Y
Supports Popular SIP Phones Y Y
Provide Client VoIP SDK(Android, iOS, Windows, macOS) Y Y
Plugin Free – WebRTC Y Y
One-click conference Y Y
Meeting Recording Y Y
Unlimited Users Y Y
Meeting Participants Included 3 100
Full REST API Y Y
Security Y Y
E164 Number processing Y Y
Agenda