2200-19000-001 Polycom SoundStation Duo Conference Phone

$485

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Part Number: 2200-19000-001
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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
    • 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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$485

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2200-19000-001 Polycom SoundStation Duo Conference Phone
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