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Grandstream GXP1450 HD IP Phone
[GXP1450] |
- Next-generation 2-line enterprise/business SIP phone
- HD audio quality and rich feature functionality
- Dual RJ-45 10/100Mbps ports with integrated PoE
- 2 line keys and 3 XML context sensitive soft keys
- HD audio handsfree speaker phone with echo cancellation
- 1 year warranty against hardware defects
- 14 day money back guarantee
- 30 day email support for technical installation issues
Product Overview
GXP1450 is a next generation enterprise grade IP phone that features 2 lines with 2 SIP accounts, a 180x60 backlit graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE, and 3-way conference. The GXP1450 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NBN/IMS platforms. It is a perfect choice for enterprise users looking for a high quality, feature rich IP phone with affordable cost.
Feature Summary
- HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance
- Automated phone book synchronization with directory server using XML. Large phonebook (up to 2,000 contacts) and call history (up to 2,000 records)
- Integrated real-time web applications (weather, stock, currency, RSS news, etc.)
- 2 line keys with dual colour LED line indicators and 2 independent SIP accounts, 3 XML dynamic context sensitive soft keys, and 5 navigation/menu/volume keys
- 10 Dedicated Feature Keys: Volume, 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, 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
Technical Summary
- 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
Hardware Summary
- IEEE 802.3af standards based Power Over Ethernet (PoE), Max power consumption 2.6W (power adapter or 3.6 (PoE)
- LAN interface: 2 x RJ45 10/100Mbps autosensing
- Headset Jack: Both RJ9 and 2.5mm jacks for professional and casual use
- LED: 2 dual colour line indicator LEDs and 1 message indicator LED
- Phone Case: ABS plastic, 32-button keypad
- Display: Back-lit 180x60 pixel backlit graphical LCD display with up to 4 level grayscale
- Universal Switching Power Adapter: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 800mA, UL certified
- Dimensions: 186mm (W) x 210mm (L) x 81mm (D)
- Handset Weight: 800g
- Package Weight: 1300g
- Temperature: 32 - 104 ºC (0 - 40 ºF)
- Humidity 10-90% (non condensing)
- Compliance: FCC / CE / C-Tick / RoHS
| Quantity |
1-4 |
5-9 |
10+ |
| Price for each |
£48.99 |
£47.77 |
£46.30 |
| Your savings |
- |
2% |
5% |
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| This product was added to our catalog on Monday 17 January, 2011. |
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