Grandstream GXP2000 Mainstream Enterprise 4-line PoE IP Phone

The Grandstream GXP2000 IP Phone is a next generation, enterprise SIP telephone that is feature rich, easy to use, supports Power-over-Ethernet and is competitively priced.

The Grandstream GXP2000 IP Phone is expandable, secure and easy to manage and offers multiple lines, multiple SIP accounts, advanced functions, superior audio quality and interoperability with most SIP end user devices, IP PBX and Softswitches. The Grandstream GXP2000 IP Phone received the prestigious "2006 Internet Telephony Excellence Award" and the "2005 Product of the Year" award from the Internet Telephony Magazine.

It is ideal for small office and enterprise customers as well as service providers.

Features

• Open Standards compatible with SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP (both client and servers), PPPoE, TFTP, NTP and TLS
• Advanced Digital Signal Processing (DSP), silence suppression, VAD, CNG, AEC and AGC giving superb audio quality
• Custom Ring-tone Software converts most music files to a custom ring tone
• Standard voice features including caller ID, call waiting, hold, transfer, forward, Intercom, shared call apperances, mute, headset, autodial, off hook dial and click to dial
• Advanced Functionality in multi-line support, multi-party conferencing, line extensions, headset enabled, TLS SRTP (pending), multi-language support (MLS) and XML enabled

Feature Keys

• 4 direct lines with 7 speed dial keys
• Up to 11 line calls (expandable to 112 lines with 2 daisy-chained Grandstream GXP2000EXT)
• 8 dedicated keys:
  • Message Button
  • Hold
  • Transfer
  • Conference
  • Speakerphone
  • Send
  • Mute/Delete
• 5 display/menu navigation keys, dual color LEDs

Telephony Features

• Intuitive graphic user interface
• Downloadble phone book (XML, LDAP)
• Muti-language supportt (MLS)
• Support customizable LCD screen via downloadable XML by HTTP/TFTP
• Dynamic negotiation of codec and voice payload length
• Call hold, call transfer (attended/blind), do-not-disturb, call forwarding, mute
• Redial, call log and volume control
• Caller ID display or block
• Call waiting, call waiting caller ID
• 3-way conferencing
• Voice mail indicator
• Downloadable custom ring tones
• In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
• Support DNS SRV look-up and SIP server fail-over
• Full-duplex hands-free speakerphone with Acoustic Echo Cancellation (AEC) and AGC
• Off Hook Auto Dial, Auto Answer, Early Dial and Speed Dial
• Adaptive jitter buffer control with packet delay and loss concealment