Polycom SoundStation IP 6000

$962.61
VoIP IP 6000 HD Conference Telephone with 1 x LAN, 220 Hz to 14 kHz audio, (AC Adapter included, PoE ready)
Enjoy clearer, more productive business conversations using this IP conference phone specifically designed for small to midsize room
Quantity

VoIP SIP 6000 HD Conference Telephone with 1 x LAN, 220 Hz to 14 kHz audio, (PoE required, AC Adapter optional). Optional power supply: PM6000PS. Will connect to PoE switch but will not work with generic universal PoE injector.
 


Features and Benefits

  • Polycom HD Voice technology for high-fidelity calls at up to 14 kHz – An industry first, an IP conference phone that sounds as natural as being there
  • Patented Polycom Acoustic Clarity™ technology – delivering the best IP conference phone experience without compromise
  • 12-foot (4-meter) microphone pickup – ideal for small to midsize conference rooms. Add even greater range with optional expansion microphones
  • Strong, robust SIP software – leveraging the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features
  • Robust VoIP interoperability – Compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration
  • Resists interference from mobile phones and other wireless devices while delivering clear voice conferencing with no distractions
  • High-resolution display – enables robust call information and multi-language support

Product Specifications

Power
• IEEE 802.3af Power over Ethernet (built in)
• Optional external universal AC power supply: 100-240V, 0.4A, 48V/19W

Display
• Size (pixels): 248 x 68 (W x H)
• White LED backlight with custom intensity control

Keypad
• Standard 12-key keypad
• Context-dependent soft keys: 3
• On-hook/Off-hook, redial, mute, volume up/down

Audio features
• Loudspeaker
• Frequency: 220-14,000 Hz
• Volume: Adjustable to 86 dB at 1/2 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
• Supported Codecs
• G.711 (A-law and Mu-law)
• G.729a (Annex B)
• G.722, G.722.1
• G.722.1C
• Siren 14

Call handling features
• Shared call / bridged line appearance
• Busy Lamp Field (BLF)
• Distinctive incoming call treatment/call waiting
• Call timer
• Call transfer, hold, divert (forward), pickup
• Called, calling, connected party information
• Local three-way conferencing
• One-touch speed dial, redial
• Call waiting
• Remote missed call notification
• Automatic off-hook call placement
• Do not disturb function

Other features
• Local feature-rich GUI
• Time and date display
• User-configurable contact directory and call history (missed, placed, and received)
• Customizable call progress tones
• Wave file support for call progress tones
• Unicode UTF-8 character support.

Multilingual user interface encompassing Simplified Chinese, Danish, Dutch, English (Canada / US / UK), French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, Swedish

Network and provisioning
• Ethernet 10/100 Base-T
• 2.5mm connection port
• EX mic ports: Two RJ-9 ports
• IP Address Configuration: DHCP and Static IP
• Time synchronization with SNTP server
• FTP / TFTP / HTTP / HTTPS serverbased central provisioning for mass deployments. Provisioning server redundancy supported.
• Web portal for individual unit configuration
• QoS Support – IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS and DSCP
• Network Address Translation (NAT) support – static
• RTCP support (RFC 1889)
• Event logging
• Local digit map
• Hardware diagnostics
• Status and statistics
• User selectable ringer tones
• Convenient volume adjustment keys
• Field upgradeable

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

Safety
• CE Mark
• EN60950-1
• IEC60950-1
• UL60950-1
• CAN/CSA C22.2 No.60950-1-03
• AS/NZS60950-1
• RoHS Compliant EMC
• FCC Part 15 (CFR 47) Class B
• ICES-003 Class B
• EN55022 Class B
• CISPR22 Class B
• AS/NZS CISPR22 Class B
• VCCI Class B
• EN22024 Telecom
• AS/ACIF S004
• Telepermit
• KCC
• GOST-R
• TRA

Protocol support
• IETF SIP (RFC 3261 and companion RFCs IEEE 802.3af Power over Ethernet version ships with
• Telephone Console
• 25 foot Ethernet cable
• Quick Start Guide
• Quick User Guide AC Power version ships with
• Telephone Console
• 25 foot Ethernet cable
• Universal Power Supply
• 7 foot region-specific power cord
• Power Insertion Cable
• Quick Start Guide
• Quick User Guide

Environmental conditions
• Operating temperature: 32 - 104 degrees F (0 - 40 degrees C)
• Relative humidity: 20%-85% (noncondensing)
• Storage temperature: -22 - 131 degrees F (-30 - 55 degrees C)

Warranty
• 1 year
• Country of Origin
• Thailand

Phone dimensions
• 14.5 x 12.25 x 2.5 in (36.8 x 31.1 x 6.4 cm) (L x W x H)

Phone console weight
• 1.75 lb (0.8 kg)

Box dimensions
• 13.0 x 15.5 x 6.0 in (33 x 39.5 x 15 cm) (L x W x H)

Box weight
• 5.1 lb (2.32 kg)

Polycom
PM6000

Data sheet

Lines
1
Ports
LAN
Conference
Yes
Headset support
2.5 mm jack
RJ9
Power
AC Adapter Included
PoE Ready
Provisioning
Ftp
Http
Https
Tftp
Codecs
G.711 (alaw)
G.711 (ulaw)
G.722
G.729a
G.729ab
Networking
QOS
VLAN 802.1Q
Other
Echo Cancel
Speaker