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Grandstream GXP2010 IP Phone
Recommended for receptionist/call centre 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
- 18 speed dial keys and 3 XML context sensitive soft keys
- 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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£88.99
£102.34 inc VAT |
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| Volume Discount: |
5+ |
£87.99
£101.19 inc VAT |
10+ |
£86.99
£100.04 inc VAT |
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| If you need larger quantities or are a reseller then please email us. |
The GXP2010 was designed by Grandstream to be the cost-effective choice for the busy call center or small business that wants the feature functionality of a key-system. The GXP2010 features include 4 direct lines with independent SIP accounts, 18 programmable speed dial keys, multiline support for up to 22 lines with LED indicators for each, 3 dynamic context sensitive XML soft keys, 4-way multi-party conferencing, a backlit 240x120 high resolution graphic LCD with multi-level grey scales, 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, 3 XML dynamic context sensitive soft keys, and 5 navigation/menu/volume keys
- 18 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
- 11 Dedicated Feature Keys: Do not Disturb (DND), Intercom, Directory , 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, 5-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 18 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: 2.5 mm and RJ11 for broad headset compatibility
- LED: 22 dual colour line indicator LEDs and 1 message indicator LED
- Phone Case: ABS plastic, 53-button keypad, comes with two detachable footstands (different heights to support different angles) and spacers for wall mounting
- Display: Blue backlit 240x120 high resolution graphic LCD with multi-level grey scales
- Universal Switching Power Adaptor: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 1200mA, UL certified
- Dimensions: 251.5mm (W) x 201.7mm (L) x 77.5mm (H)
- Handset Weight: 1106g
- Package Weight: 1761g
- 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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