2017 latest 2.8" color screen wireless voip phone with 2 rj45 1 usb port POE optional FIP11W for hotel and business use

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Overview
Quick Details
Place of Origin:
Guangdong, China (Mainland)
Brand Name:
FlyingVoice or OEM
Model Number:
FIP11W
Type:
VoIP Phone
LCD size:
2.8-inch, 320x240 -TFT color screen
LED light:
Red color LEDs for call and message indication
Management:
Support TR069 management mode
Interface:
2xRJ45, 1xRJ9, 1xFXO
Voice Codec:
G.722, G.711μ/A, G.723.1, G.726, G.729AB, iLBC
SIP:
8 SIP accounts 8 Lines
Keys:
5 navigate key,12 Dialing Buttons (0~9, *, #),32 Multi-function keys
WiFi:
802.11n
hardware:
CPU Core: MIPS 580Mhz, 16M Flash, 64M SDRAM

A high-end atmosphere very popular color wireless phone. 

FIP11WP is a new generation of enterprise-class color IP phone, 8 SIP accounts, 2.8-inch TFT color display, 8 soft keys, PoE optional, RJ9 headset. FIP11WP can display 32 speed-dial keys / BLF extension keys directly on the screen, providing convenient communications experience to call a busy corporate office and management personnel. FIP11WP excellent HD HD voice quality, rich and leading edge telephony system functions to provide users with personalized information customization, support automatic deployment, advanced security protection mechanism, compatible with a wide range of mainstream industry SIP / IMS platform.

The built-in internal antenna make user connect the network much easier and more quickly,  user can not only connect FIP11WP to the network by using a ethernet cable, but also he/ she can connect FIP11WP to the wireless network, the performance is the same as mobile phone. User can connect the wireless network on FIP11WP's LCD or on phone's webpage. Detailes about the connectons, please refer to FIP11WP user manual.



Appearance


      1.Front Panel
 
      2.Side Panel
 

 

        3.Rear Panel



Applied Environment

Operation Temperature: 0~50 Degree C
Storage Temperature: -25~ 85 Degree C
Relative Humidity: 10%~90% No Condensing
Shock: Up to 75cm (30 inches) `Drop upon Package

Codec and Voice Features

The FIP11WP supports 5 codec: G.711 (A-Law, U-Law), G.729A/AB, G.723 with 5.3 kbps and 6.3 kbps, iLBC and G.722 (HD).

The FIP11WP supports VAD (Voice Active Detection), CNG (Comfort Noise Generation), AEC, ANC (Automatic Noisy Cancel), AGC (Automatic Gain Control) and adaptive Jitter buffer.

Call Features

8 lines, Call Waiting, Auto Answer, Music on hold, Caller ID and call waiting ID, 3-way Conference, Call Hold, Call Forwarding, Call Mute, Delayed Hotline , Redial, Call Transfer: blind transfer and attended transfer, Dial Plan, Phonebook, Black List, MWI, SMS, DND,Full-duplex Speakerphone
Call log: redial list, answered calls and missed calls
Volume Adjustment: Handset/Headset, Speaker and Ringer
DTMF Relay: In-band, Out-band (RFC2833) and SIP INFO

Management

Menu Configuration, SNTP Time Synchronization, Daylight Saving Time, Alarm Clock, Password Reset, Web access management, Local and Remote Syslog (RFC3164), Factory Default, Firmware Upgradeable, Web Management Interface, Multi User Level, SNMPv2, TR069, Auto Provisioning: TFTP (including option 66), HTTP and HTTPS

Applications

MAC Address Cloning, VPN: PPTP and T2TP, DMZ, Direct IP to IP calling, IP conflict detection, Built-in 
NAT Router, DHCP Server and Client,
Layer 2 QoS: 802.1Q/VLAN ID and 802.1p PRI
Layer 3 QoS: SIP QoS, RTP QoS and Data QoS
SIP proxy redundancy: dynamic via DNS SRV, A records
NAT Traversal: Static NAT Route and Traversal by STUN
Support PoE complies with IEEE802. with3af (optional)

Web Configuration

The FIP11WP supports web configuration with MD5 authentication, and two level configurations which are admin mode and user mode. In admin mode user can configure all parameters, while in user mode user can only configure partly parameters with partly can’t configuration such as VoIP settings.

Below picture is the login web: user enters the user name and password to login and start to configuration:



Below picture is the FIP11WP web:



Explanation:

Place 1: Display the main information, such as firmware version, DSP version, Current Time, and configuration mode. Press Logout button to logout.
Place 2: Click navigation bar, and then many sub-navigation bars will appear in the place 4.
Place 3: Display the briefly explanation of the corresponding web; user can get help from it directly.
Place 4: Choose sub-navigation bars to open one configuration webpage.
Place 5: The configuration column.
Place 6: Parameters and settings value.
 
Below picture is the SIP account configuration page:


 

Explanation:

Basic
1) Line Enable--- enable the sip line;

Proxy and Registration
1) Proxy Server: enter the ip address of your sip server;
2) Outbound Server: if your sip provider provides the outbound server IP address, please enter it; Backup Outbound Server:
3) enter it when your sip provider provides this;

Subscriber Information
1) Display Name: enter the name  displayed on the phone's LCD, anything letter or number is OK;
2) Phone Number: enter the phone number your sip provider provides;
3) Account: enter the account your sip provider provides, if no, please enter the phone number or live it bank according to your sip provider's protocol;
4) Password: Enter the password of your phone number that your sip provider provides
 
The following is the wireless connection page:

 

FIP11WP Main Features

FIP11WP is high-end office IP phones. It has eight lines, four dynamic soft keys. Support 802.3af PoE (optional), high-quality speaker, 8 multi-function keys, 32 BLF, and RJ9 headset port. Each line can be configured independently of the telephone number, or composite shared telephone number.

FIP11WP support the latest configuration of various methods, such as: TR-069, HTTPS, HTTP and TFTP. Users can fully experience the high-definition voice, eight function keys, and intuitive menu options to bring convenient, fast.

FIP11WP can be configured as AP mode, at the same time acts as a IP phone, you can think it is AP .

FIP11 Series phone Features Comparison


DescriptionFIP11FIP11WFIP11WLFIP11WP
Picture
 
LED2.8-inch
color LCD screen
320 * 240 resolution
2.8-inch
color LCD screen
320 * 240 resolution
2.8-inch
color LCD screen
320 * 240 resolution
2.8-inch
color LCD screen
320 * 240 resolution
WIFIx
Line8888
Ethernet Port2*RJ45
10/100Mbps
2*RJ45
10/100Mbps
2*RJ45
10/100Mbps
2*RJ45
10/100Mbps
HeadsetRJ9RJ9RJ9RJ9
USB1111
Line Portxx1*RJ11 Fxox
Soft Key4444
 Multi-function
key
8888
POExxx
CodeG.711(A-law, U-law), G.723, G.729A/B, G.722,iLBC
TR069
SNMP
VPNPPTP/L2TP/Open VPN
Auto
configuration
             TPTP/HTTP/HTTPS Tro69 Network management
Telephone book
Call history
SMS

FIP11WP Specification

Hardware Features
Interface2 RJ-45 ports (Ethernet 10/100 Base-T)
1 RJ9 earphone jack
1 FXO port to connect PSTN line(only for L series)
1 usb port
802.11n 2T2R (Only for W series)
Support PoE comply with IEEE802.3af(optional)
Display320×240 -TFT color screen
Red color LEDs for call and message indication
Phone Keys4 soft keys
13 function keys:MENU, MUTE ,SMS, HOLD,Volume+/-,Transfer,
3-way conferencing, Hands free Redial and so on.
5 navigate key
12 Dialing Buttons (0~9, *, #),
32 Multi-function keys
Power SupplyAC/DC Adapter
  * AC Input: 100~240V, 50~60Hz
  * DC Output: 5V, 1A

Environmental 
Operation Temperature: 0~50 Degree C
Storage Temperature: -25~ 85 Degree C
Relative Humidity: 10%~90% No Condensing
Dimension18.8cm(L)*18.5cm(W)*3.7cm(H)
Unit Weight 625g (No packet)
Software Features
Audio Codec FeaturesG.711 (A-Law, μ-Law)
G.722(HD)
G.723.1
G.729AB
G.726
iLBC
Adaptive Jitter Buffer Management
Voice Activity Detection
Comfort Noise Generation
Echo Cancellation
ProtocolsSIP V2 (RFC 3261,3262,3263,3264)
SDP (RFC2327)
RTP/RTCP (RFC1889 and RFC1890)
NAPTR for SIP URI Lookup (RFC2915)
STUN (RFC 3489)
ARP/RARP (RFC 826/903)
SNTP (RFC 2030)
TFTP/HTTP/HTTPS for Auto Provisioning
DNS/DNS SRV (RFC1706 and RFC 2782)
IEEE802.1Q VLAN/802.1p and IP TOS
SNMPV2
TR069(TR104/TR098/TR111)
Management Firmware Upgradeable
Web Management Interface
Password Management
Local and Remote Syslog (RFC3164)
Auto Provisioning: TFTP, HTTP and HTTPS
SNTP Time Synchronization
Multi User Level
Auto Provision
Support TR069 management mode
ApplicationsSMS Function
MAC Address Cloning
SIP proxy redundancy
DNS SRV
Direct IP to IP calling
Traversal by STUN
VPN: pptp,l2tp,OpenVPN
QoS with Layer 2 and Layer 3
DHCP Server and Client
IP conflict detection
Support Mutil-Vlan
Call Features8 lines
Auto Answer
3-way Conference
Music on hold
Call Hold
Call Forwarding
Call Mute
Call Transfer
Call Waiting
Call Pickup
MWI
CID and CWCID
DTMF Relay: In-band, Out-band(RFC2833) and SIP Info
Full-duplex Speakerphone
Hotline
Redial
Dial Plan
Volume adjustment- handset/handfree/speaker/ringing
Phone FeaturesCustomized Ring Tone
SMS (100 records)
Call History ( 100 records )
  - Most Recently Missed Calls
  - Most Recently Received Calls
  - Most Recently Dialed Numbers
Phone book (500 records )