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GRANDSTREAM
GXV-3000
Grandstream GXV-3000 is a next generation advanced IP video phone based on SIP and H.264 standard. Built upon Grandstream's innovative technology, GXV-3000 features superb audio and video quality, rich functionality, ease of use, stylish exterior design, and highly attractive price.
Feature highlights:
- Support SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP (both client and server), PPPoE, TFTP, NTP, Telnet, TLS (pending), etc.
- Support dual 10M/100M auto-sensing Ethernet ports configurable to operate under either switch or router mode.
- Powerful video DSP with advanced adaptive jitter control and packet loss concealment technology to ensure superb audio and video quality.
- Support advanced H.264 base line real-time video codec (at CIF or QVGA resolution and up to 30 frames/second)
- Support popular voice features including 3 line indicators, full-duplex hands free speaker phone, 3-way conference, etc
- Support advanced video features including:
- High quality 5.6-inch TFT color LCD panel 2 dimensionally
rotable and camera tiltable to allow virtually all viewing angles.
- Advanced VGA resolution CMOS camera sensor
- Anti-flickering
- Auto focus and auto exposure
- Zoom
- PIP (Picture-in-Picture)
- Audio mute and camera block (for privacy)
- Still picture capture/store/send (VGA Resolution)
- Support 2 USB (2.0) host ports, 1 audio and 1 video output jack and headset jack
- Support device configuration via LCD, Web browser or central secure configuration file
GXP-2020
The GXP2020 Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price. The GXP2020 provides excellent voice clarity, a comprehensive set of advanced call features, multi-language support, security protection, automated provisioning, and broad compatibility with leading SIP platforms.
Feature highlights:
- Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
- Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
- Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
- 6 lines indicators with individual SIP account profiles
- 7 programmable hard keys and 4 XML programmable context-sensitive soft keys
- Multi-line support of up to 13 call appearance lines with dual-color LED indicators
- Expandable to additional 112 lines through expansion key-modules
- Support Caller ID display or block, per call or permanent
- Call waiting, hold, mute, transfer (blind or attended), forward, and more
- Multi-party conferencing (up to 5-way)
- And many more enterprise grade features
- Full duplex speakerphone w/advanced audio echo cancellation
- Headset jack (2.5 mm and RJ11) for broad headset compatibility
- Secure and automated provisioning for mass deployment
- SRTP and TLS (pending) for privacy protection
- And many more enterprise grade features
GXV-2010
The new GXP2010 Key System IP phone has a new sleek design and delivers excellent call quality and enterprise grade feature set that includes advanced XML capabilities, multi-party conferencing, multi-language support, presence and BLF (busy lamp field), security protection, automated provisioning, and broad compatibility with leading SIP platforms. The GXP2010 offers 4 lines, 18 programmable keys, 3 dynamic context-sensitive XML soft keys, dual switched 10M/100Mbps auto-sensing Ethernet ports with integrated PoE, and a large high-resolution backlit LCD display. As with all models in the GXP Series, this model supports comprehensive voice codecs and SRTP for privacy protection.
The GXP2010 was designed to be the cost-effective choice for the busy call center or small business that wants the feature functionality of a key-system. The 18 programmable speed-dial keys enable one-button access to office personnel and the 11 dedicated keys create one-button access to indispensable telephony features including hold, conference, mute, do-no-disturb, intercom, transfer, speaker and voicemail. This phone is ideal for enhancing office productivity and improving efficiency.
Technical Features & Benefits:
- Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
- Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
- Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
- Backlit 240x120 high resolution graphic LCD with multi-level gray scales
Call Features & Benefits:
- 4 lines indicators with individual SIP account profiles
- 18 programmable hard keys and 3 XML programmable context-sensitive soft keys
- Multi-line support of up to 22 call appearance lines with dual-color LED indicators
- Support Caller ID display or block, per call or permanent
- Call waiting, hold, mute, transfer (blind or attended), forward, and more
- Multi-party conferencing (up to 4-way) and shared line appearances
- Expandable through expansion key-modules (pending)
- And many more enterprise grade features
Handset Features & Benefits:
- Full duplex speakerphone w/advanced audio echo cancellation
- Headset jack (2.5 mm and RJ11) for broad headset compatibility
- Secure and automated provisioning for mass deployment
- SRTP and TLS (pending) for privacy protection
- And many more enterprise grade features
GXP-2000
Grandstream GXP-2000 is a next generation enterprise IP telephone based on open industry standards. Built on innovative technologies, GXP-2000 features market leading superb audio quality, rich functionalities, and excellent manageability at affordable prices.
Feature highlights:
- Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP
- Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
- Multiline support of up to 11 lines indicators (expandable to a few dozen more through expansion key-module)
- Graphical LCD to display up to 8 lines and 22 characters per line
- Dual 10/100Mbps Ethernet ports
- Headset jack
- Support Caller ID display or block, per call or permanent
- Call waiting, Hold, Mute, Transfer (blind or attended), Forward, and more
- Multi-party conferencing
- Integrated Power-over-Ether (802.3af)
- And many more enterprise grade features
GXV-1200
The GXP1200 is a very affordable 2-line entry-level SIP phone that delivers excellent sound quality and advanced feature set. The GXP1200 supports traditional telephony features such as caller id, call waiting, mute, park, hold, transfer and voicemail as well as advanced enterprise features such as 2 independent SIP accounts, multi-language support, 3-way conferencing, and custom ring-tones for distinctive call handling.
The GXP1200 also supports a broad range of voice codecs and features dual 10/100mbps auto-sensing Ethernet ports with integrated Power over Ethernet, headset jack, and secure central configuration file for mass deployment. GUI web interfaces make the GXP1200 quick and easy to install and manage. Broad interoperability with major SIP based platforms ensures compatibility with any IP communications system. Its easy installation, hands-free duplex speakerphone, backlit graphic LCD display, eight dedicated function keys, and XML applications make the GXP1200 a very cost-effective choice for any business that needs a feature rich two line IP phone.
Technical Features & Benefits:
- SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (both client and server), PPPoE, TFTP, NTP, Telnet
- Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af)
- Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more
- 128x32 pixel graphic LCD with backlight
Call Features & Benefits:
- 2 lines appearances with individual SIP account profiles and dual-color LED
- Support Caller ID display or block
- Call waiting, hold, mute, transfer (blind or attended), forward, and more
- 3-way conferencing
- 5 navigation/menu keys
- 8 dedicated buttons - hold, speakerphone, send, transfer, conference, headset, mute, messages
Handset Features & Benefits:
- Full duplex speakerphone w/advanced audio echo cancellation
- Headset jack (RJ11) for broad headset compatibility
- Secure and automated provisioning for mass deployment
- SRTP and TLS (pending) for privacy protection
- And many more enterprise grade features
GXP-280
The GXP280 is a next generation small business SIP phone that features 1 line appearance, 128x32 graphic LCD, 3 XML programmable softkeys, and dual 10M/100Mbps network ports. It is an affordable choice for a SOHO user needing a feature rich, 1-line IP phone.
Features:
- 128 x 32 pixel graphical LCD with support for multiple languages
- 1 line appearance with FLASH to handle up to 2 simultaneous calls
- 3 XML programmable context sensitive soft keys, 3 way conference
- High fidelity wideband audio, full-duplex speakerphone with advanced acoustic echo cancellation
- Dual switched auto sensing 10/100 Mbps network ports
- Automated provisioning for mass deployment, SRTP
- SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS (A record and SRV), DHCP, PPPoE, TFTP, NTP, STUN, SIMPLE
- Dual switched 10/100 Mbps
- 128 x 32 pixel graphic LCD
- 3 XML dynamic context sensitive soft keys, 3 navigation/menu/volume keys, 7 dedicated function keys for: HOLD, SPEAKERPHONE, SEND, TRANSFER, CONFERENCE, FLASH and HEADSET
- Support for G.723.1, G.729 A/B, G.711 u/a-law, G.726, G.722 (wide-band), GSM and iLBC, in- band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
- Hold, Mute, Transfer, Forward, 3-way Conference, Downloadable Phone Book (XML, LDAP up to 200 items), XML Customization of Screen, Call Waiting, Call Log, Off-hook Auto Dial, Auto Answer, Click-To-Dial, Downloadable Ringtones, Server Redundancy and Fail-over Support
BT-200
Display / hands-free IP terminals. 1 and 2 ports
Firmware upgrades via http
Easy configuration through web or keyboard/display.
Feature highlights:
- Protocols: SIP 2.0, TCP/UDP/IP, RTP/RTCP, http, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP.
- NAT supported through STUN or symmetric RTP.
- Audio codecs supported: G.723.1 (5.3 k / 6.3 k), G.729 A/B, G.711 (A-law & -law), G.726, G.728. Dynamic codec and voice payload length negotiation..
- DISPLAY with Caller ID, call waiting, call transfer, in-band & out-of-band audio DTMF (RFC2833 & INFO), dial plan, off-hook auto-dial, configurable emergency calls.
- Full-duplex hands-free, redial, call listing, volume control, voice mail indicator.
- Silence supresión supported, Voice Activity Detection (VAD), Comfort Noise Generation (CNG), Echo cancelation (G.168) and automatic gain control (AGC).
- DIGEST authentication and encryptation using MD5 & MD5-sess supported.
- Easy configuration through web or keyboard / display.
- Support for layer 2 (802.1Q VLAN, 802.1p) and layer 3 QoS (ToS, DiffServ, MPLS).
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