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VOIP SIP IP Phone with 3 way calling conference

VOIP SIP IP Phone with 3 way calling conference
MPN : SDSIP-5000
SKU : I5P
Brand :Generic
Stock : 195pcs
Price:£46.89 Exc VAT
Price:£56.27 Inc VAT
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Introduction:
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Make Free Calls and Talk for Free all Over The World:

Safecom''''s SD SIP-5000 VOIP SIP IP Phone is a cost competitive, feature-rich Ethernet LANPhone providing voice/data convergence using state-of-the-art technology. The SD SIP-5000 seamlessly integrates with IP standard-based telephony systems using the SIP protocol stack. With features such as 3-way conference call, 2concurrent calls, high quality voice and a 2 x 16 character LCD providing all the necessary information preceding a calling activity, users can experience the familiar operation of a hi-end legacy PSTN telephone. SDSIP-5000 also supports Auto Batch Provision through HTTP and TFTP which can help you to auto configure many sets simultaneously.

The Safecom''''s SD SIP-5000 IP-Phone is a fully featured Ethernet business phone that allows both business and residential customers to benefit from IP Telephony services. It reduces costs by receiving local and long distance voice services and data services over a single network connection.

The SDSIP-5000 IP-Phone is a fully-featured digital business phone that plugs directly into a customer''''s LAN and converts voice to IP right on the desktop, enabling voice and data traffic over a single wiring infrastructure on the spot and over a single packet (IP) connection back to the service provider''''s network.

 
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Features:
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Supports 2 concurrent calls
3-way conference call
Supports Auto Batch Provision/Configuration through HTTP and TFTP
Multiple Domain Registration (up to 3 different service domains)
Address book
Caller ID Display (CID)
Last number redial (LNR)
Auto redial
Speed dialing
Auto answer
Call waiting
Call forwarding
Call transfer
Call screening
Call rejection
Dialed call history
Received call history
Missed call history
Missed called indicator
Message indicator
Adjustable volume
Mute
Music on hold
Handsfree speakerphone
Headset interface
Consultative Call Transfer
Customizable ring tone
DTMF Generator
DTMF Relay: Support in-band RTP voice mixing and out-band DTMF over RTP(RFC2833)
Support SNMP
NAT & firewall bypass via STUN or pre-configured NAT gateway port mapping

 
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Voice Processing:
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Codec: G.711 (a-law and u-law), G.723.1(A) and G.729A, G.729AB.
Acoustic Echo Cancellation
Voice activity detection (VAD)

 
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Protocol:
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SIP call signaling - RFC3261 backward compatible with RFC2543RTP/RTCP
RFC1889/RFC1890)  SDP (RFC2327)
Capacity exchange based on SDP (RFC3264)
NAPTR for SIP URI lookup (RFC2915)
E.164 Number and DNS (ENUM, RFC2916)
DHCP / PPPoE (optional) for automatic IP address assignment/Static IPassignment
SNTP/TFTP for batch provision
IEEE 802.1Q VLAN
Support IP ToS and 802.1P Class-of-Service

 
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Specifications:
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Physical Specifications:

2 x RJ4510/100 Mbps Ethernet ports
2 x RJ11 ports (1 for handset, 1 for headset)
2 x 16 character display
Keypad: 12 dialing buttons (0~9, *, #), 9 fixed function buttons, 20 DSS-keyswith 2  color LED, volume up, volume down 

chipset: Agere 8300 
2Mb flash
16Mb SDRAM
Dimensions: 228 x 201 x 115mm
External power adaptor AC: 100~240V, 47~63Hz DC:9V, 1.1A
 

Environmental Specifications:

Operational temperature 0~40C
Storage temperature 0~70C
Relative humidity 95% Max. (Non-condensing)

 
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Package Contents:
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SIP Phone
Power Adapter
Phone Stand
Device for avoiding EMI (Used with Power Adaptor)
User Manual

 
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Shipping Information:
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SIP Phone with Giftbox:
Netweight: 1.3kg (IP phone Hardware only)
Gross weight: 1.5kg (phone+stand+adapter+manual+Gift box)
Carton:
Netweight : 13kg 
Gross weight : 15kg
Dimension : 44(L)cm * 36(W)cm * 50(H)cm
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