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
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
Gateway Voice features Call hold Call waiting and per call-waiting blocking Call forward (always, busy, no answer, power off) Voip speed dial Voip digital map E.164 dial plan and customize dial rules Caller number/name display Voice promote Configurable jitter buffer size Configurable audio frame Life-line function Anti-radar design Support bridge and router model In carton
The ht-322 is designed as a compact, high performance, and low cost fxo gateway. The fxo detection is optimized to avoid the hold up of the pstn line when the other party is disconnected. This has been one of the key issues in the design of fxo gateway. The incoming pstn caller id is also transmitted to the voip user for more user friendly operation. The ht-322 is a full featured fxo gateway and is designed for easy installation and configuration. It is an ideal solution for voip to pstn termination in both sme and soho environment.
The ht-912 is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. It bundles with lots of features to meet the demand in various network environments. It is an ideal low cost voip solution for travelers and soho users.
The ht-812p is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. By connecting a pstn line to the bypass port, the phone set connected to the fxs port can also access the pstn line for traditional telephone service. The ht-812p is a full featured fxs gateway and is designed for easy installation and configuration. It is an ideal low cost solution for travelers and soho users.
Supplier: Voip phone, voip gateway
Voip phone Voip phone one port/two ports Usb phone Dialup phone Gate way one port/two ports
