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Logos IP Conference Phone - Sonic HD Technology

$490.00
Tax included

Condition New

 

         IP Conference Phone dapat diregister ke IP PABX  gratis 2 buah Mic Extended

Tersedia juga tipe baru yang dapat dihubungkan ke Handphone, sehingga bisa Conference dengan GSM. Juga tersedia Conference Phone tipe analog untuk dikoneksi ke telkom Line atau Extension Analog PABX Panasonic

 LOGOS Voice Crystal IP CONFERENCE PHONE adalah  professional VoIP conference phone dengan teknologi chipset  SonicClear III HD sound yang menjamin suara jernih seperti suara asli

LOGOS VoiceCrystal IP CONFERENCE PHONE  menggunakan prinsip kerja frekuensi band yang paling lebar dengan full-duplex acoustic echo cancellation, automatic gain control, dynamic noise reduction dan  intelligent microphone mixing dengan kisaran frekuensi 200Hz-8KHz yang cocok dengan frekuensi pendengaran manusia  yang cocok dengan semua jenis  audio codec pada semua audio  dan video system. Cocok untuk telepresence meeting dari video conferencing and audio teleconferencing.

Juga tersedia IP Conference Phone. dengan penambahan konverter  ATA ( USD 67 )

SIP Accounts 3 SIP lines;

 

Segera beli di Sophos Technology

Quantity

  • Security policy (edit with Customer reassurance module) Security policy (edit with Customer reassurance module)
  • Delivery policy (edit with Customer reassurance module) Delivery policy (edit with Customer reassurance module)
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Phone Set Features

128x64 dot matrix LCD dilengkapi  LED back light;

Enriched with features 'hold', 'mute', 'forward', 'transfer', 'do not disturb', 'hand-free' keys;

LCD menu and navigation botton

 

Audio Specification

Voice frequency between  300Hz - 6999Hz; suitable for human voice frequency

Audio CODEC  G.711a, G.711u, G.729, iLBC, G.722

DTMF pass through, RFC2833 and DTMF SIP info support;

Adaptive jitter buffer;

Silence suppression and comfort noise generation;

Packet loss handling

Hi-Fi audio amplifier very strong magnet speaker with amplifying  90dB to ensure

Strong voice,  suitable for meeting room

4 built-in good quality microphones with expandable microphones to support large number of

Meeting member person at meeting room

 

Network Protocol

SIP RFC3261 fully compliant;

HTTP/FTP/TFTP for centralize automatic provisioning;

SNTP for time synchronization;

NAT traversal with periodical refreshing SIP registration.

 

 

Phone Features

Built-in web GUI for provisioning and configuration;

Data and time display;

Customer logo and text display;

Integrated call history display for incoming call, outgoing call, missed call, for tracing of missed call, dialed called, etc

Built-in phone book that is accessible from key pad.

 

Calling Features

Call hold, transfer, call forward, call pickup;

Built-in 3 way conference;

Full caller id information display;

Direct dial from call history and phone book, redial function;

Voice mail indication and retrieval;

Do not disturb function;

Call forwarding setup, forward all, busy forward, offline forward, no answer forward.                         

          

Security Features

HTTP 1.1 basic/digest authentication for web server setup;

MD5 for SIP authentication, RFC2069, RFC2617;

CBCOM based SIP encryption.

 

 

Network Interface

Integrated with 2 RJ45 Ethernet ports for LAN and WAN connections, support 802.1p/Q tagging.

 

Technology & Parameter

Recommended room: 20-60sq.m;

Microphones gather: 5.0m;

Working Temperature: 0°C-50°C;

Humidity: 20%-85%;

Speaker Volume: Max 90dBA;

Audio band-width: 300-7000Hz;

Reverberation time: <0.4S;

Noise Degree: <48dBA;

Dimensions: 305x305x64mm;

Power Requirement: AC Input: 110/220 V, 50/60 Hz, 12W; 

Power Requirement: DC Output: 18V/1A.

 

Company certifications

CE Mark (LVD & EMC directive );

RoHS.

 

Product certifications

UL, CE, FCC.

 

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