Polycom SoundStation Duo Telefone Audioconferencia 2200-19000-00

2200-19000-001 Polycom 2200-19000-001 Telefone Audioconferencia SoundStation Duo (Analógico/SIP)

Tel de Áudio Conferência Polycom IP e Analógica (SoundStation Duo)

O telefone Polycom SoundStation Duo pode ser utilizado em rede analógica e em ambiente VoIP, oferecendo a mais robusta interoperabilidade baseada em padrões da indústria. Configurar o telefone SoundStation Duo é simples. Possui uma ferramenta baseada em Web que oferece opções de configuração e facilita os upgrades online de software. O grande display do telefone oferece informações sobre a chamada, funções de telefonia e amplo suporte a múltiplos idiomas. O telefone SoundStation Duo oferece uma qualidade de voz incrivelmente nítida. Desde a tecnologia Polycom HD Voice? e o áudio full duplex, até a mais recente tecnologia de cancelamento de eco e resistência à interferência de telefones móveis e dispositivos wireless, o telefone de conferência SoundStation Duo proporciona incomparáveis experiências de conferência em grupo sem interrupções.

Power
IEEE 802.3af Power over Ethernet
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

Keypad
Standard 12-key keypad
Context-dependent soft keys: 4
On-hook/Off-hook, conference, redial, mute, volume up/down, menu, navigation keys

Audio Features
3 cardioid microphones: 200-7000 Hz
Loudspeaker frequency response: 220-7000 Hz
10ft (3m) microphone pickup
Volume: Adjustable 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
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 public switched telephone network interface
2.5mm connection port***
2 RJ9 expansion microphone ports

Network and Provisioning
IP Address Configuration: DHCP and Static IP
Time synchronization with SNTP server
FTP/TFTP/FTPS/HTTP/HTTPS server-based 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
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

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
FCC Part 68
AS/ACIF S002
AS/ACIF S004
ANATEL
Telepermit
KC
GOST-R
TRA

Protocol Support
IETF SIP (RFC 3261 and companion RFCs)

SoundStation Duo ships with the following
Telephone Console
21-ft (6.4-m) combined analog and Ethernet cable with Power Injection Module
Universal Power Supply 24V, 0.5A
7-ft (2.1-m) region-specific power cord
7-ft (2.1-m) Ethernet cable
7-ft (2.1-m) telephony cable (RJ11)
Quick Start Guide

Accessories
2 expansion microphones 200 - 7000 Hz