VoIP Phone

Detailed Product Description
SKU : VoIP-Phone-6020 Voice Features Support SIP RFC3261 Codec: G.711A/u, G.7231 high/low, G.729 Echo cancellation: Support G.168, and Hands-free can support 96ms Support Voice Gain Setting, Jitter Buffer, VAD, CNG, dual GK, call forward, and Peer to Peer NAT transverse: support STUN client, Citron, AVS SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer SIP support Pubic & Private server. Can connect to public SIP and Private SIP server at the same time SIP support dual public server DTMF: Support SIP info, DTMF Relay, RFC2833 SIP application: support Call forward/transfer/holding/waiting Call control features: Flexible dial map, support Hotline, Empty calling No. Reject service, Black list for reject authenticated call, No disturb, Caller ID. Support conference call and voice record English, Spanish, Czechoslovak alternative Network Features WAN/LAN: Support Bridge and Router model Support basic NAT and NAPT Support PPPoE for xDSL Support DHCP Client on WAN Support DHCP Server on LAN Support DNS Relay, SNTP Client, Firewall Network tools in telnet server: Including ping, trace route, telnet client Maintenance and Management Support Boot Monitor Can upgrade firmware through boot monitor Can config through Web, keypad, Telnet Can upgrade firmware and configuration file through HTTP, FTP, TFTP Support Syslog