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
Support
News

Fanvil i20S

€295.50
(€351.65 inc tax)
10 days
FANVILi20S
Please sign in to buy

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

The IP intercom Fanvil I20S is a product designed for indoor resistant to dust and water (IP54).
Although that product represents a low-cost solution, it has a number of advanced features and compatibility with most platforms (3CX, Asterisk, Broadsoft, Elastix, etc).
Fanvil I20S is compatible with the SIP standard and the IP / RTP / RTCP / SRTP protocols for voice transmission in high quality, and also allows access via call, code and RFID card (3 card are included in the package).

 

Technical features:

  • 2 SIP lines
  • Numeric keypad
  • 1 button with DSS functions
  • 1 Relay board
  • Integrated RFID card reader (up to 500 cards)
  • IP54 protection
  • 802.3af PoE (Class 3 - 6:49 ~ 12.95W) or external
  • RJ45 network ports Ethernet 10 / 100Base-Tx

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

Supporting SIP 2.0 and IMS platform of operators, and Interconnecting with the mainstream manufacturers IPPBX
Embedding BRODCOM chip, Stable and reliable
duplex hands-free, providing voice quality for carrier-class
Supporting the password to open the door 、the RFID card to open the door、the remote to control the door
A button calls to any one terminal or more terminals（RingGroup ）
Supporting visitors to leave a message, and the owner play back the message by the SIP phone
Supporting three calls or Multiparty calls (multi-party meeting) 
Once access to the network, it can be used
Providing online software upgrades and WEB management 

application
Business: Office, shopping center 
Education: University, school, stadium
Health: Hospital, laboratory, pharmaceutical plant
Logistics and Transport: Airport, car park and garage
Industry: Factory and Product Center

Technical parameters

communication protocol   Interface  
Support SIP 2.0(RFC3261)and Relevant RFC power supply mode 12V+-15%/1A DC or PoE
agreement   PoE 802.3af (Class 0 - 12.95W) 48V/380mA
Speed dial button The buttons adopt stainless steel，sturdy and durable LAN 10/100BASE-TX s Auto-MDIX, RJ-45）
Speed dial button number one The recommended cable five or more
Digital keys Adopt Touch Button Passive switch Often open and close, maximum achievable 30V/1A AC/DC
Voice   The active switch output 12V/500mA DC
microphone one The structural characteristics  
Speaker 1W Shell Material surface cover Aluminum,plastic pan
Hands-free and full-duplex support Working temperature 0°C to 55°C
Audio stream   Environmental relative humidity 10% - 95%
Support agreement RTP Storage Temperature -40°C to 70°C
CODEC G.711,G.729,G.722,GSM,G.726,iLBC installation Pit mounted（pre-bury ）
RFID   Dimension overall dimensions：174.5╳96╳44mm
EM-40XX(125KHz)   Installation dimensions：148╳80╳36mm
 Supporting SIP 2.0 and IMS platform of operators, and Interconnecting with the mainstream manufacturers IPPBX
 Embedding BRODCOM chip, Stable and reliable
 duplex hands-free, providing voice quality for carrier-class
 Supporting the password to open the door 、the RFID card to open the door、the remote to control the door





  A button calls to any one terminal or more terminals (RingGroup ）
 Supporting visitors to leave a message, and the owner play back the message by the SIP phone
 Supporting three calls or Multiparty calls (multi-party meeting) 
 Once access to the network, it can be used
 Providing online software upgrades and WEB management 

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