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Grandstream GXP1610

Grandstream GXP1610

SKU: GXP1610


The Grandstream GXP1610 is a simple IP phone for a small-to-medium businesses (SMBs) or home office use. This Linux-based model features single SIP account, up to 2 call appearances, and 3 XML programmable soft keys. A 132x48 LCD screen creates a clear display for easy viewing. Additional features such as dual switched 10/100 Mbps ports, multi-language support, call-waiting, and 3-way conferencing allow the GXP1610 to be a high quality, user-friendly basic, versatile and dependable office phone.

  • Next-generation single line enterprise/business SIP phone
  • Dual RJ-45 10/100Mbps ports
  • 3 XML context sensitive soft keys
  • Clear 132 x 48 pixel LCD display
  • 3-way conferencing
  • 2 year warranty against hardware defects
  • 14 day money back guarantee
  • 30 day email support for technical installation issues
  • Full-duplex hands-free speakerphone with advanced acoustic echo cancellation
  • Automated phone book synchronization with directory server using XML. Large phonebook (up to 500 contacts) and call history (up to 200 records)
  • Integrated real-time web applications (weather, stock, currency, RSS news, etc.)
  • 3 XML dynamic context sensitive soft keys, and 5 navigation/menu/volume keys
  • Voice Features: Downloadable Phone Book (XML, LDAP), XML Customisation of Screen, Voice Mail Indicator, Redial, Call Log, Volume Control, Caller ID Display or Block, Call Waiting, Hold, Transfer, Forward, FLASH, Mute, 3-Way Conferencing, off-hook auto dial, configurable emergency dialling (e.g., 911), early dial, click-to-dial, auto answer, downloadable ring tones
  • Dual RJ-45 10/100 autosensing ports (switched or routed) to allow a desk PC to be connected to the phone in order to conserve switch/router ports and AC sockets in order to simplify deployment
  • RJ9 headset jack (allowing EHS with Plantronics headsets)
  • NAT router and DHCP server
  • Multi language support (English, German, Italian, French, Spanish, etc.)
  • Secure Real-time Transport Protocol (SRTP) and SIP Transport Layer Security (TLS) (pending) support
  • Full duplex handsfree speakerphone with Advanced Echo Cancellation (AEC), Acoustic Gain Control (AGC) and side tone support
  • Easy configuration through manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
  • NAT-friendly remote firmware upgrade capability via tftp/http even from behind firewalls/NATs
  • Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products (e.g. 3CX, Asterisk®/Trixbox® IP PBX, MS Messenger, Cisco® IP phone and gateway, etc)
  • Dynamic negotiation of codec and voice payload length
  • SIP and DNS server redundancy and failover
  • Features Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
  • Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology
SKU: GXP1610
10 available

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