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Call the Headset Help Line: 800-457-4287 |
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SoundStation Duo Conference Phone
CP-SSDUO
| List Price: |
$849.00 |
| Price: |
$791.00
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| You Save: |
$58.00 (6%) |
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Use in analog or IP mode and keep it up-to-date with simple online software upgrades
Compatible with a broad array of IP call platforms
Circuit switched telephony support and migration to SIP-based platforms
Broadest interoperability with leading SIP-based IP PBX and Softswitch platforms
24/7 reliability with automatic failover from SIP to circuit switched
Lower cost of deployment and administration
Outstanding performance at an affordable price
Legendary audio conferencing experience and performance
Crystal clear group audio conferencing
with HD voice in IP mode
Exceptional room coverage and mic pickup
Expandable coverage area with optional expansion microphones
Easy to use with proven UI and industrial design
Auto failover from IP to analog and failback for continuous operation in case of a network failure
Unparalleled voice clarity via Polycom HD Voice technology
Easy to deploy and administer with a web configuration tool that eliminates the need for a boot server
Superior call handing, security, and provisioning, the SSDUO leverages the most advanced IP endpoint software in the industry
Connect to mobile phones and PCs for internet dialing
The phone’s large backlit display offers easily visible call information, telephone functions, and broad multi-language support
IEEE 802.3af Power over Ethernet or optional external universal AC power supply: 100-240V, 24V, 0.5A, 2.5mm DC plug
Display size (pixels): 248 x 68 (W x H)
White LED backlight with custom intensity control
Standard 12-key keypad, 4 context-dependent soft keys, on-hook/off-hook, conference, redial, mute, volume up/down, menu, navigation keys
3 cardioid microphones: 200-7000 Hz, loudspeaker frequency response: 220-7000 Hz, 10ft (3m) microphone pickup
Adjust volume to 86 dB at 0.5 meter peak volume
Individual volume settings with visual feedback for each audio path
Voice activity detection
Comfort noise fill
DTMF tone generation/DTMF event RTP payload
Low-delay audio packet transmission
Adaptive jitter buffers
Packet loss concealment
Acoustic echo cancellation
Background noise suppression
SIP call handling features: call hold, call transfer, divert (forward) and pickup,
distinctive incoming call treatment/call waiting, advanced local three-way conferencing, one-touch speed dial, redial remote missed call notification, and automatic off-hook call placement
1 year manufacturer's warranty
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