Polycom SoundStation Duo IP & Analog Conference Phone has Broad Connection Options
- Manufacture Discontinued 01/22 – no direct replacement
- 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 manufacture warranty
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