2200-19000-001 Polycom SoundStation Duo Conference Phone
$485
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Features Of 2200-19000-001
- Crystal clear conversations featuring HD Voice and echo cancellation
- 3 built-in cardioid microphones with up to 10 foot pickup range
- Can add optional external expansion microphones for additional coverage
- 248 x 28 pixel white LED backlit display
- Supports both analog and IP telephony platforms – Ethernet 10/100 BaseT port for SIP deployments; Two-wire RJ-11 analog port for PBX or PSTN deployments
- 24×7 reliability with automatic failover from IP to analog
- Interoperable with leading SIP-based PBX and softswitch platforms
- Easy administration with no boot server required
- Powered via 802.3af Power over Ethernet (PoE) when used with direct network connection in SIP mode
- Powered with included Power Injection Module (PIM) when used in PSTN mode or dual PSTN/SIP mode
Specifications Of 2200-19000-001
- Power
- IEEE 802.3af Power over Ethernet
- External universal AC power supply: 100–240 V, 24 V, 0.5 A, 2.5 mm DC plug
Display
- Size (W x H): 248 x 68 pixels
- White LED backlight with custom intensity control
Keypad
- Standard 12-key keypad
- Context-dependent soft keys: 4
- On-hook/Off-hook, conference, redial, mute, volume up/down, menu, 5-way navigation keys
Audio Features
- 3 cardioid microphones: 200–7000 Hz
- Loudspeaker frequency response: 220–7000 Hz
- 10 ft (3 m) microphone pickup
- Volume
- Adjustable to 86 dB at 0.5 meter
peak volume
- Adjustable to 86 dB at 0.5 meter
- Full-duplex
- Type 1 compliant with IEEE 1329
- 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
- Supported codecs
- G.711 (A-law and Mu-law)
- G.729a (Annex B)
- G.722
- iLBC 13.33 and 15.2kbps
SIP Call Handling Features
- Call hold*
- Call transfer, divert (forward) and pickup
- Distinctive incoming call treatment/call waiting
- Advanced Local three-way conferencing (conference, join, split, hold, resume)
- One-touch speed dial, redial*
- Remote missed call notification
- Automatic off-hook call placement
- SIP URI dialing
- Do not disturb function
- Shared call/bridged line appearance
- Busy Lamp Field (BLF)
- Multicast Group Paging and Push-to-Talk
Other Features
- Automated failover (SIP to PSTN)
- SIP Server Redundancy
- Time and date display/call timer
- User-configurable contact directory and call history (missed, placed, and received)
- Corporate Directory (LDAP) support
- User selectable ringer tones
- Wave file support for call progress tones
- Unicode UTF-8 character support
- Multilingual user interface encompassing Simplified Chinese, Traditional Chinese Danish, Dutch, English (Canada/US/UK), French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, Swedish
- Called, connected party information
- Support for multiple Caller ID standards**
- Bellcore Type 1
- ETSI
- DTMF
Interfaces
- Ethernet 10/100 Base-T
- Two-wire RJ-11 analog PBX or PSTN interface
- 2.5 mm connection port***
- 2 RJ9 ports for wired expansion microphones
Network and Provisioning
- IP Address Configuration
- DHCP and Static IP
- Time synchronization with SNTP server
- FTP/TFTP/FTPS/HTTP/HTTPS serverbased central provisioning for mass deployments. Provisioning server redundancy supported.
- Web portal for individual unit configuration and online software upgrade
- QoS Support—IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP
- Telchemy® VQmon® support
- Network Address Translation (NAT) support—static
- RTCP support (RFC 1889)
- Configuration import/export
- Local digit map (dialing plan)
- Hardware diagnostics
- Status and statistics
- Reset to factory settings
Security
- Transport Layer Security (TLS)
- Encrypted configuration files
- Digest authentication
- Password login
- Support for URL syntax with password for boot server
- HTTPS secure provisioning
- Support for signed software executables
- IEEE 802.1x Network Access Control
Protocol Support
- IETF SIP (RFC 3261 and companion RFCs)
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