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Grandstream GXP2000 IP Phone
Recommended for mainstream enterprise/business use
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- Next-generation 4-line enterprise/business SIP phone
- Superb audio quality and rich feature functionality
- Dual RJ-45 10/100Mbps ports with integrated PoE
- 7 programmable speed dial keys with LED line indicators
- Handsfree speaker phone with advanced echo cancellation
- 1 year warranty against hardware defects
- 14 day money back guarantee
- 30 day email support for technical installation issues
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| Please select PSU type required: |
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£65.99
£75.89 inc VAT |
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| Volume Discount: |
5+ |
£64.99
£74.74 inc VAT |
10+ |
£63.99
£73.59 inc VAT |
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| If you need larger quantities or are a reseller then please email us. |
Independent Reviews/Awards
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Recommended - The Grandstream GXP-2000 IP phone is a good choice if you need multiple account handling flexibility and extension phone capabilities, for example, in a small business or home office. |
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Winner of 2006 Internet Telephony® "Excellence" award - The GXP2000 IP Phone has excelled in the VoIP/IP Telephony industry, and most importantly, its customers are willing to offer their testaments of support. |
The Grandstream GXP2000 next-generation SIP VoIP Phone offers market-leading superb sound quality and rich functionality and is ideal for mainstream business/enterprise deployment. The GXP2000 features include 4 direct lines with independent SIP accounts, 7 programmable speed dial buttons (up to 112 additional speed dial buttons can be provided using the GXP2000 Extension Module), multiline support for up to 11 lines with LED indicators for each, 8 line 22 character LCD display with blue backlight, dual switched 10/100 autosensing network ports with integrated PoE, and a full duplex speakerphone with advanced acoustic echo cancellation.
- 4 SIP line keys with dual colour LED line indicators and 5 navigation/menu/volume keys
- 7 programmable speed dial keys (for call pickup/forward/transfer) with dual colour LED indicators, LEDs can be configured as Busy Lamp Fields (BLF) to indicate if a particular extension is busy with a call or idle
- 8 Dedicated Feature Keys: Message, Hold, Transfer, Conference, Headset, Speakerphone, Redial/Send, Mute/Del
- 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) with integrated PoE 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
- 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., 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
- SIP Presence package (RFC3856, 3863) for use of 7 programmable line keys, SIP MESSAGE method (RFC3428) for up to 100 incoming IM messages, SIP PUBLISH method (RFC3906), SIP dialogue package (RFC4235)
- IEEE 802.3af standards based Power Over Ethernet (PoE) support
- Voice codecs supported include G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.722 (wide-band), G.726 (32K), GSM and iLEB codecs.
- Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
- In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
- Support for Fixed IP, DHCP, SIP Presence (SIMPLE), Automated NAT traversal using IETF STUN (manual configuration of firewall/NAT not required) and symmetric RTP (compatible with Cisco’s ATA-186, etc)
- Protocols supported include SIP RFC3261, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (client and server), NTP, TFTP, PPPoE
- Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
- DIGEST authentication and encryption using MD5 and MD5-sess
- LAN interface: 2 x RJ45 10/100Mbps autosensing
- Headset Jack: 2.5mm Headset port
- LED: 11 dual colour line indicator LEDs and 1 message indicator LED
- Phone Case: ABS plastic, 35-button keypad
- Display: 130x64 pixel graphic LCD (8 lines of 22 characters) with blue backlight, can be tilted to adjust viewing angle
- Universal Switching Power Adaptor: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 1200mA, UL certified
- Dimensions: 220mm (W) x 215mm (L) x 57mm (H)
- Handset Weight: 880g
- Package Weight: 1355g
- Temperature: 32 - 104 ºC (0 - 40 ºF)
- Humidity 10-90% (non condensing)
- Compliance: FCC / CE / C-Tick / RoHS
Asterisk® (www.asterisk.org) is registered trademark of Digium, Inc. Trixbox® (www.trixbox.org) is a registered trademark of Fonality, Inc. Novavox Limited is not affiliated with, nor endorsed by either of the companies listed above.
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