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Fanvil X4G

110.00
(130.90 inc tax)
30 days
FANVILX4G
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The IP phone Fanvil X4G is a professional product ideal for companies. It allows, thanks to a dual-display system, to enter up to 30 DSS / BLF customizable functions.

 

It 'a SIP standards compatible product with 4 configurable account, dual color display, power supply via PoE and dual network port Gigabit 10/100/1000 Mbps.

 

Specifications:

  • 4 SIP Account configurable
  • color LCD 2.8 "(320x240)
  • Color display for dedicated DSS controls
  • 6 DSS buttons with tricolor LEDs (up to 30 configurable BLF)
  • Function phone book (500 contacts)
  • Black Colour
  • Power via PoE
  • 2 Gigabit network ports 10/100/1000 Mbps
  • Headphone jack RJ9
  • Handset (HS) / Hands-free (HF) / Headphone (HP) (EHS Supported for Plantronics headsets)
  • Provisioning via FTP / TFTP / HTTP / HTTPS / DHXP OPT66 / SIP PNP / TR069

 

 

 

Note: the power adapter is not included in the package and must be purchased separately.

 

Generic
 4 SIP Lines
 HD Voice
 POE Enabled
 2 LCDs (Main + DSS)
 Handset(HS) / Hands-free(HF) / Headphone(HP) mode(EHS support for Plantronics headsets)
 Intelligent DSS Keys
 Desktop / Wall-mount installation
 Optional External Power Supply Economical and Environmental friendly package
 Industrial Standard Certifications: CE/FCC

Call Features
 Call out / answer / reject
 Mute / Unmute (microphone)
 Call Hold / Resume
 Call Waiting
 Intercom
 Caller ID Display
 Speed Dial
 Anonymous Call (Hide Caller ID)
 Call Forwarding (Always/Busy/No Answer)
 Call Transfer (Attended/Unattended)
 Call Parking/Pick-up (depending on server)
 Redial/Auto-Redial
 Do-Not-Disturb (per line / per phone)
 Auto-Answering (per line)
 Voice Message (on server)
 Local 3-way Conference
 Hot Line
 Hot-Desking

Phone Features
 Phonebook (500 entries)
 Remote Phonebook (XML/LDAP)
 Call log (100 entries, in/out/missed)
 Black/White List Call Filtering
 Voice Message Waiting Indication (VMWI) 
 Programmable DSS/Soft keys
 Network Time Synchronization
 Action URL / Active URI

Audio
 HD Voice Microphone/Speaker (Handset/Hands-free, 0 ~ 7KHz Frequency Response)
 Wideband ADC/DAC 16KHz Sampling
 Narrowband CODEC: G.711a/u, G.723.1, G.726-32K, G.729AB
 Wideband CODEC: G.722
 Full-duplex Acoustic Echo Canceller (AEC) – Hands-free Mode, 96ms tail-length
 Voice Activity Detection (VAD) / Comfort Noise Generation (CNG) /Background Noise Estimation (BNE) / Noise Reduction (NR)
 Packet Loss Concealment (PLC)
 Dynamic Adaptive Jitter Buffer up to 300ms
 DTMF: In-band, Out-of-Band – DTMF-Relay(RFC2833) / SIP INFO

Networking
 Physical: 10/100Mbps(X4), 10/100/1000Mbps(X4G) Ethernet, dual bridged port for PC bypass
 IP Configuration: Static / DHCP / PPPoE
 Network Access Control: 802.1x
 VPN: L2TP (Basic Unencrypted) / OpenVPN
 VLAN
 QoS

Protocols
 SIP2.0 over UDP/TCP/TLS
 RTP/RTCP/SRTP
 STUN
 DHCP
 PPPoE
 802.1x
 L2TP (Basic Unencrypted)
 OpenVPN
 SNTP
 FTP/TFTP
 HTTP/HTTPS
 TR069

Deployment & Maintenance 
 Auto-Provisioning via FTP/TFTP/HTTP/HTTPS/DHCP OPT66/SIP PNP/TR069
 Web Management Portal
 Web-based Packet Dump
 Configuration Export / Import
 Phonebook Import/Export
 Firmware Upgrade
 Syslog

Physical Specifications
 Main LCD x1: 2.8 inch (320x240) color-screen LCD
 DSS Key-mapping LCD x1:color-screen LCD
 Keypad: 38 keys, including
4 Soft-keys
6 Function keys
5 Navigation keys
12 Standard Phone Digits keys
3 Volume Control keys, Up/Down/Mute(Microphone)
1 Hands-free key
6 DSS Keys with tri-color LED
1 Page-Jump/Configuration (PJC) key
 HD Hands-free Speaker (0 ~ 7KHz) x1
 HD Hands-free Microphone (0 ~ 7KHz) x1
 HD Handset (RJ9) x1
 Standard RJ9 Handset Wire x1
 1.5M CAT5 Ethernet Cable x1
 RJ9 Phone Jacket x2: Handset x1, Headphone x1
 RJ45 Ethernet Jacket x2: Network x1 (802.3AF POE Class 2 Enabled), 
PC x1 (Bridged to Network)
 Main Chipset: Broadcom
 DC Power Input: 5V/1A
 Power Consumption: (X4) Idle – ~ 1.3W, Peak – ~4.3W
(X4G) Idle – ~ 1.7W, Peak – ~5.7W
 Working Temperature: 0 ~ 40℃
 Working Humidity: 10 ~ 65%
 Dual-Functional Back Rack x1: Desktop Stand / Wall-mount
 Color: Black
 Package Dimensions: 285x270x65mm (W x H x L)
Generic
 4 SIP Lines
 HD Voice
 POE Enabled
 2 LCDs (Main + DSS)
 Handset(HS) / Hands-free(HF) / Headphone(HP) mode(EHS support for Plantronics headsets)
 Intelligent DSS Keys
 Desktop / Wall-mount installation
 Optional External Power Supply Economical and Environmental friendly package
 Industrial Standard Certifications: CE/FCC

Call Features
 Call out / answer / reject
 Mute / Unmute (microphone)
 Call Hold / Resume
 Call Waiting
 Intercom
 Caller ID Display
 Speed Dial
 Anonymous Call (Hide Caller ID)
 Call Forwarding (Always/Busy/No Answer)
 Call Transfer (Attended/Unattended)
 Call Parking/Pick-up (depending on server)
 Redial/Auto-Redial
 Do-Not-Disturb (per line / per phone)
 Auto-Answering (per line)
 Voice Message (on server)
 Local 3-way Conference
 Hot Line
 Hot-Desking

Phone Features
 Phonebook (500 entries)
 Remote Phonebook (XML/LDAP)
 Call log (100 entries, in/out/missed)
 Black/White List Call Filtering
 Voice Message Waiting Indication (VMWI) 
 Programmable DSS/Soft keys
 Network Time Synchronization
 Action URL / Active URI

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