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High Definition Color PoE Gigabit IP Phone

#VIP-1260PT

฿3,900.00 | In stock 5

Models:

Description

Intuitive Design Brings Quality Communication

PLANET VIP-1260PT is a new six-line SIP enterprise Gigabit IP phone with 2.8" LCD display that brings lifelike richness and voice quality to phone calls. The VIP-1260PT is a reliable communication device as it features ease of use, versatility, high-quality design and superb audio performance. Besides, its full duplex speakerphone system with HD voice can definitely make communication between two or more business parties crystal-clear with no noise interference in the background as it supports the G.722 wideband and Opus codec.

 

Multi-language IP Phone with Color Display

As the VIP-1260PT is a high-performance IP phone, it allows you to make calls from any location easily as long as it is online. Its 2.8-inch (main) LCD monitor comes with multi-language support via screen and web UI. It complies with IPv4/IPv6, IEEE 802.3af/at PoE interface and dual 10/100/1000Mbps Ethernet for flexible deployment and supports superior audio quality delivered by the advanced speaker and microphone system, and the digital signal processor (DSP).

 

 

Compliant with SIP 2.0

SIP phones continue to gain popularity among businesses as the preferred protocol for enhancing communication across IP networks. The VIP-1260PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-1260PT is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better voice over IP services.

 

Affordable for All Businesses

The VIP-1260PT is definitely affordable for all business establishments who want flexible deployment options and expansion. It can effortlessly deliver secure toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service) and IP ToS technology.

 

High-quality G.722 HD and Opus Audio Codes 

The VIP-1260PT delivers with Harman Kardon speaker, wideband G.722 HD and opus audio codec whose both hardware and software HD functions (HD speaker and G.722 audio codec) are the next generation of voice quality for telephony audio, making the quality of voice better than that (toll quality) of the standard digital telephony and come close to that of a room conversation. HD voice is transmitted in the audio frequency range of 50Hz to 7KHz or higher over telephone lines, resulting in higher quality voice and clearer communication. The VIP-1260PT keeps bringing the most premium sound for the users.

Specification

Hardware
Lines (Direct Numbers)6 SIP Lines
DisplayLCD x 1: 2.8 inch (320 x 240) color-screen LCD
Feature KeysKeypad: 31 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
Network InterfacesRJ45 10/100/1000 Mbps Ethernet jacket x 2:
Network x 1 (802.3af PoE Class 2 enabled)
PC x 1 (Bridged Network)
ConnectorsHD hands-free speaker (0 ~ 7KHz) x 1
HD hands-free microphone (0 ~ 7KHz) x 1
HD handset (RJ9) x 1
Power RequirementsIEEE 802.3af
5V (Optional External Power Supply)
Power Consumption: Idle – ~ 1.7W, Peak – ~5.7W
Weight700g
Dimensions (W x D x H)180 x 210 x 80 mm
Protocols and Standard
ProtocolsSIP2.0 over UDP/TCP/TLS
RTP/RTCP/SRTP
STUN
DHCP
IPv6
LLDP
PPPoE
802.1x
L2TP (basic unencryption)
OpenVPN
SNTP
FTP/TFTP
HTTP/HTTPS
TR069
AES128 & AES256
Networking
NetworkingPhysical: 10/100/1000Mbps Ethernet, dual bridged port for PC bypass
IP Model: IPv4/VIPv6/VIPv4&VIPv6
IP Configuration: Static, DHCP, PPPoE
Network Access Control: 802.1x
VPN: L2TP (basic unencryption), OpenVPN
VLAN
QoS
RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)
Deployment & MaintenanceAuto-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
Features
Call FeaturesCall 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
Do-Not-Disturb (per line, per phone)
Auto-Answering (per line)
Voice Message (on server)
Local 3-way Conference
Hot Line
Hot-Desking
Phone FeaturesIntelligent phonebook (up to 1000 entries in total)
Remote phonebook (XML/LDAP, 1000 entries)
Call log (100 entries in total, in/out/missed)
Black/White List Call Filtering
Screen saver
Voice Message Waiting Indication (VMWI)
Programmable DSS/Soft keys
Network Time Synchronization
Voice Recording with IP PBX
Action URL / Active URI
Multi-language support in screen and web UI: English, Chinese (Traditional/Simplified), Japanese, Russian, Italian, Turkish, German, Dutch, Spanish, Hebrew, Polish, French, etc.
Audio FeaturesHD voice microphone/speaker (handset/hands-free, 0~7KHz frequency response)
HAC handset
Wideband ADC/DAC 16KHz sampling
Narrowband codec: G.711a/u, G.726-32K, G.729AB, iLBC
Wideband codec: G.722, Opus
Full-duplex acoustic echo canceller (AEC)
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
Environment
Operating Temperature0 ~ 45 degrees C
Operating Humidity10 ~ 95% (non-condensing)
EmissionCE, FCC, RoHS

Solution

Enhanced, Full-Featured SIP IP Phone

The VIP-1260PT is optimized for executive use for administrative assistants and those working with bandwidth-intensive application on collocated PCs. Four programmable extension keys could be configured as IP PBX features like BLF, MWI, DND, Call Forward, Call Park and many others.

 

Enterprise IP Telephony Deployment of VIP-1260PT

Help your teams move faster with exceptional audio quality and built-in flexibility. The VIP-1260PT is much easier to install and configure than the traditional phone system. Its low cost and high-definition voice quality give you value for money. Based on standard SIP 2.0, it is compatible with all the standard SIP-based servers.

Feature

Highlights

  • IETE SIP compliant with Transport Layer Security
  • Wideband G.722 HD and Opus audio
  • 2.8-inch 320 x 240 color LCD monitor
  • 6 identities/accounts with 2-line color LED key 
  • Dual Gigabit and IEEE 802.3af/at PoE compliant
  • Multi-language support via the web UI and LCD
  • IPv6, VPN, VLAN, QoS, TR069 and auto-provisioning

 

Advantageous Applications

  • Supports SIP 2.0 (RFC3261) over UDP/TCP/TLS
  • Inband, SIP info, RFC 2833 DTMF relay
  • Soft keys and function keys programmable
  • Echo cancellation: Supports G.168, and a maximum filter length of 96ms in hands-free mode
  • Supports voice gain setting, voice activity detection (VAD) and comfort noise generation (CNG) 
  • Full duplex hands-free speakerphone
  • Hands-free headset ringing choice
  • Voice codec setting for each SIP line

 

SIP Applications

  • Call forward and transfer (blind/attended)
  • Call holding and waiting
  • 3-way conferencing
  • Paging and intercom
  • Call park, call pickup and join call
  • Redial and click to dial
  • Secondary dialing automatically
  • Incoming calls, outgoing calls and missing calls (Each supports 100 records.)
  • SMS and speed dial
  • Phonebook up to 1000 records

 

Call Control Features

  • Flexible dial map, hotline and empty calling no. for rejected service
  • Black list for rejected authenticated calls
  • White list and call limit
  • Do not disturb (DND)
  • Caller ID display
  • Dial without registration

 

Network Features

  • IPv4/IPv6
  • Static /PPPoE/DHCP client 
  • 802.1 VLAN (voice VLAN/data VLAN)
  • VPN (L2TP) and openVPN
  • Quality of Service

 

Maintenance and Management

  • Integrated web server provides web-based administration and configuration
  • Telephone keypad configuration via display menu/navigation
  • Automated provisioning and upgrade via HTTPS, HTTP, TFTP
  • User authentication for configuration pages
  • SNTP time synchronization
  • TR069

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