HD Voice 2 SIP lines IP Phone,VoIP Phone.Asterisk Elastix mini slip Telephone RJ09 Headset interface,Multi Language support

SPECIFICATIONSprotocol: SIPis_customized: YesType: VoIP PhoneSMS and Speed dial: YesSIP lines: YesPhonebook: YesOrigin: Mainland ChinaModel Number: F52Headset interface: RJ09Color: blackCertification: CEBrand Name: Uxeitoo   HD Voice 2 SIP lines IP Phone,VoIP Phone.Asterisk Elastix SIP Phone,mini slip VoIP Phone,Telephone VoIP Related Products: Note: All our products are origninally manuafactured.100% new, any fake products, we promise fullly refund.Original is highly quality guaranteed,and we provide lifetime free tech support,also will test 24 hours to make sure it's all functional before delivery. Products Description F52 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. F52 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.   F52 supports SIP protocols, support Bridge, support Router, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service. Specifications Support SIP2.0 (RFC3261) and correlative RFCs SIP support 2 SIP servers, and backup SIP proxy servers Supports SIP UDP / TCP/ TLS SIP support SIP domain, SIP authentication (none, basic, MD5),DNS name of server, Peer to Peer / IP Call Large dot matrix LCD display and soft keys make use easier to us Soft keys programmable Function keys programmable (each key have 2-color led indication) Supports RJ-9 headset Multi languages displayed HD voice: HD handset Codec: G.711A/u, G.7231 high/low, G.729AB, G.722, G.726-32 DTMF: In-band, RFC 2833 and SIP INFO Full duplex hands- free speaker phone with acoustic echo cancellation Voice Activity Detection (VAD) Comfort Noise Generation (CNG) Background noise suppression Adaptive jitter buffers Packet loss concealment SRTP mute WEB ,telnet and LCD menu management Management with different account right Recover firmware and factory reset through POST mode Upgrade firmware via HTTP, HTTPS, FTP, TFTP DHCP option66 and custom option for auto provisioning Auto Provisioning to upgrade configuration file with AES(256)encryption TR-069 and Syslog Features Call forwarding(busy/no answer/always) Call transfer(unattended/attended/semi-attended) Call holding/waiting 3 way call conference Capable of 10 way conversation BLF list Join call Call pickup/park/back Redial Call completion Hot desking SIP messaging, MWI Flexible dial plan Barring function for outgoing calls Do not disturb Auto answer Caller ID display CLIR(rejects anonymous calls) CLIP(to make an anonymous call) Dial without registration Call logs with missed calls/ incoming calls/outgoing calls.Each support 300 records Phonebook 500 records Speed Dial(abbreviated dialing)Hotline/Warm-line Call rejection Black List/ White List Intercom/Intercom barge Password dial,DTMF hidden Action URL/active URI Voice codec setting for each SIP line Multicast LDAP,XML phonebook Web dial Keypad lock,Emergency call WAN/LAN:10/100M Ethernet ports Bridge mode and route PPPoE for xDSL 802.3af POE(optional) VLAN(Voice VLAN/Data VLAN) 802.1X NAT (NAPT) NAT transverse:Support STUN client DHCP client on WAN/ DHCP server on LAN Main DNS and secondary DNS server DNS Relay SNTP Client Firewall and DMZ Web management access restricted VPN(L2TP/Open VPN)   Hardware Adapter Input:AC 100-240V Adapter Output: DC 5V/1A WAN Port-10/100 BASE-T RJ-45 for LAN LAN Port-10/100 BASE-T RJ-45 for PC Headset:RJ9 Power consumption: IDLE:2.5W Active:2.8W Display Resolution: 128*48 pixels Operation Temperature: 0~40Features Call forwarding(busy/no answer/always) Call transfer(unattended/attended/semi-attended) Call holding/waiting 3 way call conference Capable of 10 way conversation BLF list Join call Call pickup/park/back Redial Call completion Hot desking SIP messaging, MWI Flexible dial plan Barring function for outgoing calls Do not disturb Auto answer Caller ID display CLIR(rejects anonymous calls) CLIP(to make an anonymous call) Dial without registration Call logs with missed calls/ incoming calls/outgoing calls.Each support 300 records Phonebook 500 records Speed Dial(abbreviated dialing)Hotline/Warm-line Call rejection Black List/ White List Intercom/Intercom barge Password dial,DTMF hidden Action URL/active URI Voice codec setting for each SIP line Multicast LDAP,XML phonebook Web dial Keypad lock,Emergency call WAN/LAN:10/100M Ethernet ports Bridge mode and route PPPoE for xDSL 802.3af POE(optional) VLAN(Voice VLAN/Data VLAN) 802.1X NAT (NAPT) NAT transverse:Support STUN client DHCP client on WAN/ DHCP server on LAN Main DNS and secondary DNS server DNS Relay SNTP Client Firewall and DMZ Web management access restricted VPN(L2TP/Open VPN)   Hardware   Adapter Input:AC 100-240V Adapter Output: DC 5V/1A WAN Port-10/100 BASE-T RJ-45 for LAN LAN Port-10/100 BASE-T RJ-45 for PC Headset:RJ9 Power consumption: IDLE:2.5W Active:2.8W Display Resolution: 128*48 pixels Operation Temperature: 0~40 degree Relative Humidity:10~65% Main Chipset : Broadcom SDRAM:16MB FLASH:4MB Weight-Phone:0.99kg Dimension:235*225*97 mm   Language Support   English German French Italian Spanish Hungarian Polish Spanish(South America)   Relative Humidity:10~65% Main Chipset : Broadcom SDRAM:16MB FLASH:4MB Weight-Phone:0.99kg Dimension:235*225*97 mm   Language Support English German French Italian Spanish Hungarian Polish Spanish(South America)