1, Not need to register SIP PROXY, supports Point-to-Point communication (call the other partyí»s IP address directly);
2, Not need to connect any network cable, supports mutual communication between two FXS ports;
3, Three SIP PROXY and accounts registered at the same time for each FXS port (so total is 6 user names), one SIP PROXY offline, other one can replace it to work timely;
4, Built-in Router;
5. With reverse-polarity signal, billing function supported;
6. Support Voice-announcing IP address and operating by hand or by web page;
7, Support SIP 2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, STUN, TFTP;
8, Powerful digital signal processing (DSP) to ensure superb audio quality;
9, Advanced adaptive jitter control and packet loss concealment technology;
10, Support various vocoders including G.711 (a-law and u-law), G.723.1 (5.3K/6.3K), G.726, G.729A/B and iLBC;
11, Support Caller ID/Name display or block, Hold, Call Waiting, Flash, Call Transfer (blind/consultative );
12, Call forward, in-band DTMF, Dial Plans;
13, Support call waiting caller ID;
14, Support fax pass through and T.38 (pending);
15, Support silence suppression, VAD (Voice Activity Detection), CNG (Comfort Nois Generation);
16, Line echo cancellation (G.168), and AGC (Automatic Gain Control);
17, Support standard encryption and autherntication (DIGEST using Md5 andMD5-sess);
18, Support for Layer 2 (802.1q VLAN, 82.1P) and Layer 3 Qos (Tos, Diffserv, MPLS);
19, Support automated NAT traversal without manual manipulaton of firewall/NAT; 20, Support device configuration via built-in IVR, web browser or downloading encrypted configuration file through TFTP or HTTP;
21, Support firmware upgrade via TFTP or HTTP;
22, Ultra compact (wallet size) and lightweight design, great companion for travelers;
23, Desk/Wall mounts. Interface: WAN: 1 unit, connect to XDSL/Cable Modem or Network Switch or Router LAN: 1 unit, connect to PC or other network equipment FXS: 2 units, connect to Normal phone, DECT/Cordless phone, Electrograph, Switch etc Power: 1unit, connect to Power adapter: 12V DC---1A Products
Product characteristics:
1. Standard sip 2. 0 terminal product, powerful applicability, and best for global popularizing; support sip 2. 0, tcp / udp / ip, rtp / rtcp, icmp, arp / rarp, ntp, pppoe, tftp, http, etc; support g. 711 (alaw and u-law) , g. 723. 1 (5. 3k / 6. 3k) , g. 723 (40k / 32k / 24k / 16k) , g. 728, g. 729a / b, ilbc;
2. The powerful dsp digital audio processor to keep a wonderful audio-quality; inside dns, router, nat, dmz, the gateway and dhcp (the client end and the server end) ; jitter control adn packet loss concealment technology;
3. Support incoming call on show, restricting, and holding, disconnection, call transfer, call divert, dialing project, etc. Support conference call;
4. Supports vad, cng, agc, the echo eliminates (g. 168) and the static sound suppression;
5. Support layer 2 (802. 1q vlan, 802. 1p) and layer 3 (qos, , tos diffserv, mpls)
6. Powerful long-distance tftp or http manages; to dispose through the browser or the voice hint;
7. Nat automatic penetration;
8. Support passing through and t. 38 fax;
9. Good for the environment, according to the international electromagnetism radicalizations standard;
10. Select high quality material, original electrocircuit design, and no electric current noise, work steadily;
11. Porcelain and legerity design, let you feel the full-bodied appetency of the product besides expediency schlepping.
Ata (sip2. 0) yes
Router / bridge / dhcp yes
Ethernet port 1wan / 1lan (rj45)
Fxs 1fxs
Fxo pstn pass-through
Exit yes
Weight 0. 38kg
Active status temperature 0~40
Power source adapter ac in: 100v-240v dc out: + 9v / 600ma
Use power: <2. 5w
Authentication fcc / ce
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
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
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.
