Description
IP conference phone with broadband (HD Voice) 14kHz audio with support for SIP protocol. Designed for medium-sized rooms, and a group of people, which allows users to comfortably call up to a distance of four meters from the unit. Graphic display with backlight and high resolution. Port for wired cable connection to a PC or mobile phone. The new Polycom technology for protection from interference signal of a mobile phone. Built-in support for Power over Ethernet (PoE), remote web management. Patented technology for the degradation of surrounding noise, noise and feedback. AC power adapter included.
Features:
- IP phone SIP standard, groups of up to around 15 people
- IP conference phone with wideband HD audio and support for SIP
- Suitable for medium-sized room with up to 10 people
- When using the optional external microphone kit, the range increases up to 15 people
- 3 high-quality microphones covering around 3600 with a range of up to 4m
- A graphical backlit display (248 x 68 pixels) for easy access to functions
- 2.5 mm jack port for connecting a PC or a mobile phone
- Built-in support for Power over Ethernet (PoE), remote graphical Web Administration
- Patented technology that automatically resolves working with sound
- Removal of unwanted sounds from the environment, intelligent management microphones
- Built-in protection against mobile phone interference
Technical information:
- Power Supply: IEEE 802.3af Power over Ethernet (built-in)
- Optional external AC power source: • 100- 240V, 0.4A, 48V / 19W
- Display: Size (pixels): 248 x 68, White LED backlight with selectable brightness nestavitelným
- Keyboard: Standard 12-button, Soft keys: 3, On-hook / off-hook, redial, mute, volume up / down
- Calling a raised / lifting the handset, redial, mute, volume +/-
- Frequency: 220-22,000 Hz, Volume: Adjustable to 85 dB at 50 cm
- Individual audio settings with vyzuelní response for each channel
- Noise cancellation, DTMF
- Supported codecs: G.711 (A-law and Mu-law), G.729a (Annex B), G.722, G.722.1, G.722.1C and Siren 14
- Sharing call bridging Call
- Resolved distinction incoming calls / Call waiting
- Pointer time, call forwarding, transfer, hold, call control using the menu
- Local three-way conference, speed dialing and call forwarding one button
- Customizable contact list and call history (missed, dialed, received)
- Network: Ethernet 10/100 Base-T
- Ports: 2.5 mm jack, EX ports for microphones: 2 x RJ-9
- Setting the IP address: DHCP and Static IP
- Time synchronization with SNTP
- Network Address Translation (NAT)
- RTCP support (RFC 1889)
- Choice Ringtones
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