SDSIP-5000 VOIP SIP IP Phone is a cost competitive, feature-rich Ethernet LAN Phone providing voice/data convergence using state-of-the-art technology. The SDSIP-5000 seamlessly integrates with IP standard-based telephony systems using the SIP protocol stack. With features such as 3-way conference call, 2 concurrent 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 SDSIP-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.
Different ITSP Settings:-
SDSIP-5000 IP SIP Phone Settings for voiptalk
SDSIP-5000 IP SIP Phone Settings for gossiptel
SDSIP-5000 IP SIP Phone Settings for iptel
SDSIP-5000 IP SIP Phone Settings for sipgate
SDSIP-5000 IP SIP Phone Settings for voipfone
SDSIP-5000 IP SIP Phone Settings for voipfone and SIP Discount
Features:-
* 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
Voice Processing:-
* Codec: G.711 (a-law and u-law), G.723.1(A) and G. 729A, G.729AB.
* Acoustic Echo Cancellation
* Voice activity detection (VAD)
Protocol:-
* SIP call signaling - RFC3261 backward compatible with RFC2543 RTP/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 IP assignment
* SNTP/TFTP for batch provision
* IEEE 802.1Q VLAN
* Support IP ToS and 802.1P Class-of-Service
Specifications:-
Physical Specifications:-
* 2 x RJ45 10/100 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-keys with 2
colour 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)
Package Contents:-
* SIP Phone
* Power Adapter
* Phone Stand
* Device for avoiding EMI (Used with Power Adaptor)
* User Manual
