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IR307
is a next generation entry-level SIP Phone that features 6 SIP account profiles, high resolution dot matrix LCD display with LED back-light and programmable soft-keys. The IR307 delivers superior audio quality, comprehensive telephony features, automated provisioning, built-in encrypted SIP signaling transmission for security protection and seamless interoperability with IP PBX and leading SIP telephony platforms. IR307 is the choice for any business IP phone needs.

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Rich Phone Features - 6 SIP account profiles ; Support Caller ID display or block ; Call waiting, hold, mute, transfer, forward, etc. ; 3-way full duplex hands-free conferencing ; Built in buttons - hold, speakerphone, transfer, conference, headset, mute, messages, volume ; 4 softkey buttons.


Voice Quality - IR307 is built on a full-duplex hands-free high-definition voice engine. The GIPS® voice engine processes Net Package Jitter Buffer, AEC and many voice codecs and supports high quality full-duplex hands-free calls.


Performance & Stability - IR307's SIP Stack and the 400MHz Analog Devices BlackFin® BF531 Processor provide efficient performance with low CPU & memory resource requirement. Platform expandibility enables users to customize and add innovative applications to the phone.


Call Control
  • Basic Call (Voice)
  • Caller ID Display
  • Redial , Speed Dial, Hotline
  • Call From History & Address Book
  • Call Hold & Consultation Hold
  • Call Transfer (Blind & Attend)
  • Call Waiting
  • Call Forward (On Busy / No Answer, Unconditional, Offline)
  • 3-Way Talk
  • Automatic Accept Call / Do Not Disturb

Other Functions
  • Mute, Volume Adjust & Ring Tone Profile Management
  • Presence
  • Instant Message
  • DTMF (Inbound / Outbound / SIP-Info)
  • MWI for Voice Mail
  • Call History (Missed, Received & Dialed)
  • Phone Book (LCD Menu & Web)
  • Voice Recording & Voice Mail
  • Session Timer
  • Dial Plan & Speed Key
  • Multi-Account Management
  • Alarm (Once & Repeat)
  • SIP Stack and Call Control Logging
  • Syslog , Telnet, Ping & Trace route Function
  • 4 Softkeys

Voice Engine
  • Full Duplex Hands-free Talking
  • Acoustic Echo Cancellation (AEC),
    Voice Activation Detection (VAD),
    Comfort Noise Generation (CNG)
  • Volume Adjustment for Ringing, Speaker & Handset Individually
  • Hands-free and Headset Mode
  • Support for G.711 u/a, G.729ab & iLBC Codec
  • Adaptive Jitter Buffers and Packet Loss Concealment
  • 3 Ways Talk Mixing

 Physical Specifications
  • H x L x W:108 mm x 210 mm x 185 mm
  • Power Adapter: AC Input 100~240V, 50-60Hz, DC Output 5V, 1.2A
  • Network Interface: IEEE 802.3 10/100 Base-T
  • Power Consumption: 1.8W ~ 2.1W

Interface
  • Speed Dial Keys
  • High resolution 64*128 dot LCD
  • Handset and Hands-free Speaker
  • Two 10/100 Base-T Ethernet Ports with Bridge / Router Function
  • Web Management
  • Load Profile and Menu from XML

Protocol Features
  • SIP RFC3261 and most RFC Extensions
  • DNS (A, SRV, NAPTR)
  • STUN
  • SNMP v2 or TR069

Network
  • Manual Configuration / DHCP / PPPoE
  • Time and Date Synchronization Using NTP / Register
  • XML-Based Auto-Provision and Auto-Update
  • Support for IEEE802.1p/Q tagging (VLAN)
  • Work on Router / Bridge Mode
  • NAT Keep Alive
  • 3 layers DSCP

Interoperability Testing
  • ZTE Soft-switch and IMS Server
  • Ericsson IMS Server
  • Nokia IMS Server
  • Asterisk SIP Server
  • SER SIP Server
  • More than 30 Servers and Terminals...

 

 

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