Warning Cookies are used on this site to provide the best user experience. If you continue, we assume that you agree to receive cookies from this site. OK

Fanvil D900

(374.30 inc tax)
10 days
Please sign in to buy

This product cannot be added to the
cart because you are not logged in.

The Fanvil D900 is a pioneer for the next generation of smart video phones. Running on the Android OS 4.2 platform and built with the latest Freescale video chipset, the D900 video IP phone truly offers an intelligent OS, superior video quality and rich applications at an affordable price.
 6 SIP accounts
 The ability to simultaneously login to all server
 Possibility of registering a domain or implementation of SIP calls peer to peer
 HD voice

 Supports SIP 2.0 (RFC3261) and correlative RFCs
 SIP supports 6 SIP servers, and backup SIP proxy servers
 Supports HDMI
 Supports 3.5mm headset
 Supports USB Host
 Echo cancellation: Acoustic echo cancellation (AEC) can reach 128ms max filter length in duplex speaker phone mode
 Supports PLC & adaptive jitter buffer
 Supports HD speaker phone

 7” TFT 800x480

Feature Keys
 Capacitive touch screen
 Support multi-touch (max - 5 touch points)

Audio Features
 HD voice: HD codec, HD speaker
 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 and H.263
 Image codec: JPEG, PNG, BMP, GIF
 Video format: MP4,3GP AND FLV
 Video call resolution: QCIF/ CIF/ VGA/ 4CIF (720P comming soon)
 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 6 video display mode

Call Features
 Call forwarding
 Call transfer (blind/attended/alert)
 Call holding
 Call waiting
 Call conference
 BLF List
 Capable of 10 way conversation
 Join call
 Call completion
 Hot desk function
 Soft DSS key(max 96)
 Auto Redial / unredial
 Support multi line and predial
 Support messaging and MWI
 Voice recording during talking /local
 Flexible dial plan
 Call rejection

Phone features
 Android 4.2 OS

 Supports high quality video call
 Supports capacitive touch screen (multi-touch) Black List
 Barring function for outgoing calls
 Do not disturb
 Auto answer (handfree/ headset)
 Caller ID display
 CLIR (rejects anonymous calls)
 CLIP (to make an anonymous call)
 Dial without registration
 Supports call logs with missed calls/incoming calls/outgoing calls
 Intercom/Intercom barge
 Password dial
 Headset ring
 Direct IP call without SIP proxy
Network/ Security Features
 WAN/LAN: support bridge mode
 Supports PoE (802.3af)
 Supports main DNS and secondary DNS server
 Supports VLAN
 Support SNTP Client
 Support VPN L2TP, PPTP , IPSec
 Support SIP SRTP,
 Support web HTTP, HTTPS
 Support QoS:802.1p/q, DSCP
 Support MD5 authentication
 Support Web Filter
 DHCP/ static/ PPPoE

Maintenance & Management
 Android 4.2
 Supports third party android applications
 Supports desktop Google search
 Supports desktop widgets
 Supports Web, Telnet
 Web Management with different account right
 Supports automatic upgrades/ configuration deployment
 Supports encrypted configuration files download with AE
 Support 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
 Headset Port- 3.5mm
 HDMI Port- A type
 SD card- Optional
 USB Port- USB 2.0
 Power Consumption- Idle: 2.5W/Active: 5W
 LCD Size Diagonal:7 inch (800 x 480) Capacitive touch screen
 Camera- 5 mega pixel
 Operation Temperature 0~40℃
 Relative Humidity 10~65%
 CPU ARM Cortex –A9 (Freescale i.MX6)
 CPU Frequence-1.0 GHz Quad/Dual
 Flash- 4GB
SD card- Optional
 Weight- Phone:1.2Kg/Total:1.8Kg

 Support 2.0 and correlative RFCs
 6 SIP accounts
 The ability to simultaneously login to all server
 Voice Codecs: G.711a / u, G.7231 high / low, G.729a / b, G.722, G.726, iLBC, AMR-NB; AMR-WB
 Video codecs: H.263, H.264, MPEG4, Video resolution: QCIF, CIF, QVGA 

 Photo Format: PEG GIF and BMP PNG
 Video formats: MP4, 3GP and FLV
 Possibility of registering a domain or implementation of SIP calls peer to peer
 Can be used for custom ring tones, Support for STUN
 Support for the SIP SMS
 HD voice

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
Security Y Y
E164 Number processing Y Y