SIP/IAX2 IP Phone|VoIP Phone for Asterisk
-Model: UTP-102 -WAN: 1 X 10/100Mpbs RJ45 port -LAN: 1 X 10/100Mpbs RJ45 port -LCD display: 2 X 16 characters with back light -Speaker: 8 Ohm/0.2 Watt speaker for speaker operation -Standard: SIP RFC 3261, RFC 2543, (supports up to 2 SIP domains); IAX2 -Voice codec: G. 723.1 (5.3k/6.3k), G. 729A/B, G. 711(A-law/u-law) -Voice standard: Auto Gain Control (AGC) G. 168/165 compliant 16ms echo cancellation Auto Echo Cancellation (AEC) Voice Activity Detection (VAD) Comfort Noise Generation (CNG) In-band, out-of-band DTMF relay, RFC2833, SIP info Adaptive Jitter Buffer -Call features: Call forward, call transfer, 3-way conference call Customized dial rules Caller ID DND (do not disturb), Black List, Limited List Call history: Incoming call, Outgoing call, Missed call -Protocols: TCP/IP, DHCP, PPPoE, SNTP, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP -Applications: NAT Traversal (Outbound proxy and STUN) VPN (L2TP) -Access mode: DHCP, Static IP, PPPoE (for ADSL, Cable modem) -Management: Web, Keypad, Telnet Password protected Firmware upgrade through HTTP, FTP, TFTP Configuration values export/import Auto-provisioning -Regulatory compliant: CE, FCC part 15 class B |
Post Time:10 Nov, 2007 15:15Expiry Date:09 Nov, 2008
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