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Grandstream GXP1620 HD IP Phone
[GXP1620] |
- Next-generation 2-line enterprise/business SIP phone
- HD audio quality and rich feature functionality
- Dual RJ-45 10/100Mbps ports
- 2 dual-colour line keys and 3 XML context sensitive soft keys
- HD handset with support for wideband audio
- 2 year warranty against hardware defects
- 14 day money back guarantee
- 30 day email support for technical installation issues
Product Overview
The Grandstream GXP1620 is Grandstream’s standard level IP phone for small businesses. This Linux-based model features 2 lines, 3 XML programmable soft keys, HD audio and 3-way conferencing. A 132x48 LCD screen creates a clear display for easy viewing and the GXP1620 includes dual 10/100mb network ports. Additional features such as multi-language support, Electronic Hook Switch support for Plantronics headsets and call-waiting help make the GXP1620 a high quality, user-friendly and dependable IP phone.
Feature Summary
- HD wideband audio handset
- Full-duplex hands-free speakerphone with advanced acoustic echo cancellation
- Automated phone book synchronization with directory server using XML. Large phonebook (up to 500 contacts) and call history (up to 200 records)
- Integrated real-time web applications (weather, stock, currency, RSS news, etc.)
- 2 dual-colour 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
- 13 dedicated function keys for MUTE, HEADSET, TRANSFER, CONFERENCE, SEND and REDIAL, SPEAKERPHONE, VOLUME, PHONEBOOK, MESSAGE, HOLD, PAGE/INTERCOM, RECORD, HOME
- 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) 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
- RJ9 headset jack (allowing EHS with Plantronics headsets)
- 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. 3CX, 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 2 programmable line keys, SIP MESSAGE method (RFC3428) for up to 100 incoming IM messages, SIP PUBLISH method (RFC3906), SIP dialogue package (RFC4235)
- 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
- LAN interface: 2 x RJ45 10/100Mbps autosensing
- Headset Jack: RJ9
- LED: 2 dual colour line indicator LEDs and 1 message indicator LED
- Display: 132x48 pixel graphical LCD display
- Universal Switching Power Adapter: Input: 100-240VAC; 50-60 Hz, Output: +5VDC, 600mA
- Dimensions: 209mm (L) x 184.5mm (W) x 76.2mm (H) (with handset)
- Handset Weight: 0.73KG
- Package Weight: 1.1KG
- Temperature: 32 - 104 ºC (0 - 40 ºF)
- Humidity 10-90% (non condensing)
- FCC: Part 15 (CFR 47) Class B, CE : EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1RCM: AS/ACIF S004; AS/NZS CISPR22/24; AS/NZS 60950; AS/NZS 60950.1
| Quantity |
1-4 |
5-9 |
10+ |
| Price for each |
£38.00 |
£37.00 |
£36.00 |
| Your savings |
- |
3% |
5% |
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| This product was added to our catalog on Monday 03 September, 2012. |
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