Polycom SoundStation Duo Conference Phone Price in Qatar

Key Features

Support for analog & IP telephony platforms
Automatic failover from IP to analog
Easy administration with no boot server required

SKU: SoundStation Duo Category: Polycom IP Phone

Product Description

The Polycom SoundStation Duo conference phone offers flexibility for small to medium-sized rooms, enabling you to use existing analog lines and VoIP lines. Failover also allows you to still use your analog connection if your internet connection fails as a result of technical issues with your provider.

Polycom SoundStation Duo Key Features

Support for both analog and IP telephony platforms
Interoperable with leading SIP-based PBX and softswitch platforms
24×7 reliability with automatic failover from IP to analog
Easy administration with no boot server required
Up to 10ft (3 meter) microphone pickup range.

Best-in-class Investment Protection
The Polycom SoundStation Duo conference phone operates in analog telephony environments, however it also supports the migration to VoIP.

Mobile Compatibility
Connects to mobile phones for dialing without the need for an analog phone line. This device can also be connected to a PC or tablet to become a high-quality conference phone for Internet calling.

Polycom SoundStation Duo Conference Phone Technical Specifications

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 ἀle support for call progress tones
Unicode UTF-8 character support
Multilingual user interface encompassing Simpliἀed 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 synchronisation 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 ἀles
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

EMC

Telecom

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-speciἀc 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

Environmental Conditions

Operating temperature: 32° – 104° F (0° – 40° C)
Relative humidity: 20%-85% (non-condensing)
Storage temperature: -22° – 131°F (-30° – 55°C)

Warranty

1 year

Country of Origin

China

Phone Dimensions

15.6 x 12.9 x 2.5 in (34.6 x 32.7 x 6.4 cm) (L x W x H)

Phone Console Weight

1.62 lb (0.74 kg)

Box Dimensions

13.9 x 18 x 3.7 in (35.3 x 45.8 x 9.4 cm) (L x W x H)

Box Weight

4.56 lb (2.06 kg)

* Also available in PSTN mode

** Due to the diversity of Caller ID standards, some features may not be available in all areas. In addition, the quality of the telephone line connection may affect Caller ID functionality. Caller ID service may require a subscription from a service provider in your area.

*** SoundStation Duo uses a cable that connects to a standard 2.5mm headset connector. If your mobile phone model or PC does not support this type of connection you will need an adapter (not included).

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