VI2006 Voip Phone with 2 sip line

VI2006 is ideal for both the small office and general enterprise users. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.


VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


VI2006 supports IAX2&SIP protocols,support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.


VI2006 supports traditional telephony features such as Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.

Specifications
Support SIP 2.0 (RFC3261) and correlative RFCs
Codec:G.711 A/U Law, G.723.1, G.729a/b,G.722,G.722.1, G.726
Echo cancellation: Support G.168, and Hands-free
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speaker phone
NAT transverse: support STUN client
SIP support SIP domain, SIP authentication
DNS name of server, Peer to Peer/ IP call SIP support Pubic &Private server

Can connect to Public SIP and Private SIP server atthe same time

DTMF Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward transfer / holding /waiting

Call control features: Flexible dial map, Hotline, Empty calling reject,Black list for reject authenticated call, limit call, No disturb, Caller ID


Support three way conference call
Support voice record, 240 seconds max or 3 record max, user-defined record prompt with 1 minute max

Support Phonebook 500 records
Incoming calls / Outgoing calls / Missing calls. Each support 100 records

Support conference and voice record on SIP server 8 kind of ring type 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 VLAN (voice vlan/data vlan)
QoS with DiffServ
Support DMZSupport VPN (L2TP/UDP TUNNEL)
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route,telnet client



Maintenance and Management Web ,telnet and keypad management
Management with different account right
Upgrade firmware through POST mode
Upgrade firmware through HTTP, FTP or TFTP
Telnet remote management
Upload/download settingfileSafe mode provide reliability
Supoort Auto Provisioning (upgrade firmware orconfiguration file)
Support Syslog

VI2006 is ideal for both the small office and general enterprise users. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.


VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


VI2006 supports IAX2&SIP protocols,support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.


VI2006 supports traditional telephony features such as Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.

Specifications
Support SIP 2.0 (RFC3261) and correlative RFCs
Codec:G.711 A/U Law, G.723.1, G.729a/b,G.722,G.722.1, G.726
Echo cancellation: Support G.168, and Hands-free
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speaker phone
NAT transverse: support STUN client
SIP support SIP domain, SIP authentication
DNS name of server, Peer to Peer/ IP call SIP support Pubic &Private server

Can connect to Public SIP and Private SIP server atthe same time

DTMF Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward transfer / holding /waiting

Call control features: Flexible dial map, Hotline, Empty calling reject,Black list for reject authenticated call, limit call, No disturb, Caller ID


Support three way conference call
Support voice record, 240 seconds max or 3 record max, user-defined record prompt with 1 minute max

Support Phonebook 500 records
Incoming calls / Outgoing calls / Missing calls. Each support 100 records

Support conference and voice record on SIP server 8 kind of ring type 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 VLAN (voice vlan/data vlan)
QoS with DiffServ
Support DMZSupport VPN (L2TP/UDP TUNNEL)
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route,telnet client



Maintenance and Management Web ,telnet and keypad management
Management with different account right
Upgrade firmware through POST mode
Upgrade firmware through HTTP, FTP or TFTP
Telnet remote management
Upload/download settingfileSafe mode provide reliability
Supoort Auto Provisioning (upgrade firmware orconfiguration file)
Support Syslog

Created by: amelia, Last modification: Tue 08 of May, 2012 (07:32 UTC) by admin
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