We have detected that you are visiting our website from (). To select an alternative currency use the drop down list at the top right hand corner of the page. All currency conversions are approximate.
|
Grandstream GXP1200 IP Phone
Recommended as an entry level enterprise/business phone
|
- Next-generation entry-level dual line enterprise SIP phone
- Superb audio quality and rich feature functionality
- Dual RJ-45 10/100Mbps ports with integrated PoE
- XML programmable 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
|
| Please select PSU type required: |
|
|
|
£58.99
£67.84 inc VAT |
|
|
| Volume Discount: |
5+ |
£57.99
£66.69 inc VAT |
10+ |
£56.99
£65.54 inc VAT |
| |
| If you need larger quantities or are a reseller then please email us. |
The Grandstream GXP1200 next-generation entry-level SIP VoIP Phone offers the same market-leading superb sound quality and rich functionality as the award winning GXP2000 . The GXP1200 features include 2 line appearance with dual coloured LED indicator and 2 independent SIP accounts, 128x32 pixel screen with blue backlight, XML programmable context sensitive soft keys, dual switched 10/100 autosensing network ports with integrated PoE, and a full duplex speakerphone with advanced acoustic echo cancellation.
- 2 SIP line indicator keys and mute button, 3 XML programmable context sensitive soft keys, 5 navigation/menu/volume keys
- 8 Dedicated Feature Keys: Message, Hold, Transfer, Conference, Headset, Speakerphone, Redial/Send, Mute/Del
- Voice Features: Downloadable Phone Book (XML, LDAP up to 200 items), 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
- 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: RJ-11 Headset port
- LED: 1 dual coloured LED, 2 line indicators and mute button
- Phone Case: ABS plastic, 30-button keypad, comes with detachable footstand and spacers for wall mounting
- Display: 128x32 pixel graphic LCD with blue backlight
- Universal Switching Power Adaptor: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 1200mA, UL certified
- Dimensions (excluding stand): 195mm (W) x 201.7mm (L) x 77.5mm (H)
- Handset Weight (excluding stand): 730g
- Package Weight: 1325g
- 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.
|