JR-900 IP phone is an internet based voice network phone terminal. PHONE series IP phone 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.
JR-900 IP phone supports IAX2,SIP,protocols,Support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interface and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.
Features
Support SIP 2.0 (RFC3261) and correlative RFCs
Supprt IAX2
Codec:G.711A/u, G.7231 high/low, G.729, G.722
Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
SIP support Pubic & Private server. Can connect to Public SIP and Private SIP server at the 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 (optional: voice vlan/data vlan)
QoS with DiffServ
Support DMZ
Support 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 setting file
Safe mode provide reliability
Supoort Auto Provisioning (upgrade firmware or configuration file)
Support Syslog
10PCS/CARTON
VOIP phone Voip phone Voice features Call hold; Unattended/ attended call transfer Call waiting and per call-waiting blocking Call forward (always, busy, no answer, power off) Voip speed dial Voip digital map User define ring tone E.164 dial plan and customize dial rules Caller number/name display Voice promote Configurable jitter buffer size Configurable audio frame Data features Static/dynamic wan-ip-addressing Pppoe Management and provisioning Http, telnet, keypad and dedicate manage tool Adjustable user password and super password Url based ftp, http and tftp auto provision Configure file download and upload by palmtool Support four settings storage, switch by keypad Safe mode provide reliability Four servers store and load by keypad 80 entries each for missed calls, answered calls and dialed calls 100 entries for speed dial number Multiply language support Interfaces 2 x lan In carton
1. Ideal for group conference, free internet call and music enjoyment. 2. Due to 48KHz sample rate, presenting CD-quality music and high fidelity sound. 3. HI-FI amplifier and high-quality speaker. 4. Full-duplex conversation, crystal clear voice. 5. DSP built-in for echo cancellation and noise reduction. 6. Adjustable volume and mute control. 7. Convenient for private conversations with ear-phone. 8. Ideal for travel because of small size and light weight. 9. USB 1.0 & 2.0 compatible, No driver required. 10. Mac and PC compatible. 11. Supporting all softphone, like Skype, MSN, Yahoo messenger, AOL, Vonage and VoIPBuster . Hardware: cable, ADSL.) 1 pc/color box,20pcs/ctn GW:7.5KG NW:6.5KG CTN size:380*380*295mm
Voip phone with fxo Two separated uart for cpu and dsp32-bit risc cpu with 125mhz clock rate16-bit fixed-point dsp with 125mhz clock ratemax. Support 32mbytes sdram and 16mbytes flash Support 1 line for voip call, sip(rfc2543, arfc3261)phonebook with 100 groups memory Network protocoltftp client/dhcp client/ pppoe clienttelnet/http server dns clientiptcpudp rtp rtcpipicmparprarpsntp Voice qualityvad: voice activity detectioncng: comfortable noise generatolec: line echo cancellerpacket loss compensationadaptive jitter buffer Codecg.711: 64k bit/s (pcm)g.723.1: 6.3k / 5.3k bit/sg.726: 16k / 24k / 32k / 40k bit/s (adpcm)g.729a 8k bit/s (cs-acelp)g.729b: adds vad & cng to g.729 Call functioncall holdcall waitingcall forwardcaller id3-way conference Dtmf functionin-band dtmfout-of band dtmfsip info Sip serverregistrar server (three sip account)outbound proxy A phone functionvolume adjustmentspeed dial, phone bookflashspeaker phone Ip assignmentstatic ip and dhcppppoe support for adsl or cable modem G securityhttp 1.1 basic/digest authentication for web setupmd5 for sip authentication (rfc2069/ rfc 2617) Voip sip phone +nxd-804support poe key features: Ieee 802.3af compliant input interface Complete input / output isolation Efficiency: 78% minimalover load & short circuit protection Without fxo setting.
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Sc-2007-ie intercom ip phone is well developed according to customers requirements based on voip phone. It has fashion classic aluminum wire drawing surface shell with the design of waterproof, dustproof, anti-dismantle. The traits of strong compatibility and supporting extra access swiping card device make it perfectly to be used in bank, subway, power plant, mine, prison, steel companies and so on. It is the best choice and highly cost effective for customers' projects requirements in this industry. Qty/ctn: 20 pcs N.W/ctn: 10.4kg G.W/ctn: 13kg Carton meas: 365*365*270mm
Voip gateway Ieee 802.3 10 base tsupport dhcp, pppoe dns telnet Support g.711a/ug.723.1, g.729 /a/b/ab and gsm610 voice codec Sip, h.323 v4 protocalsip rfc3261support http, https, ftp upgrade Tcp/ip: internet protocalsupport rtp rtcp protocal Support vad/cngone wan, one lan, support router Call forward (always, busy, no answer), call waiting and per call-waiting blocking Caller number/name display caller id /block outgoing caller-id/reject anonymous call A voice activity detection (vad) facsimile protocol(option): t.30/t.38 Voip speed dial, voip digital mapper area ring tone supportdnd(do not disturb) mwi G configurable/adaptive jitter buffer sizesupport vlan protocalsupport web, ivr to configur
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