Specifications
FXS:1 support ITU H323V4 and IETF SIPV2
2 ensure high quality speech
3 support VLAN and QoS
4 NAT Transversal
5 Router
2 Channel FXS sip gateway free roaming
Product Introduction:
The two channels FXS VoIP gateway is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS Gateway). It comes with 2 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications. The HT-922 is a full featured FXS gateway and is designed for easy installation and configuration. It supports the two most widely used Open VoIP Standards (SIP and H.323). This allows the HT-922 to interoperate seamlessly with softswitches or IP PBXs made by various vendors. Its high performance offers toll quality voice, flexible networking, and feature-rich call functions. It is an ideal low cost solution for SME environment where multiple lines are required.
Key Features - LINUX OS
- Built-in HTTP Web Server
- PPPOE Dial-up
- NAT Broadband Router Functions
- DHCP Client
- Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
- Two 10/100 Ethernet for WAN / LAN connections
- Peer-to-Peer IP Calls
- Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
- Line Echo Cancellation
- VLAN and QoS support
- NAT Transversal and Router functions
- Voice prompts, HTTP Web, Auto Provision support for configuration and updates
- Highly stable embedded Linux operating system in high performance ARM 9 Processor
|
Basic Features - One RJ-11 FXS port for traditional phone set or PBX's trunk line
- LEDs for Power, Ready, Status, WAN, PC, FXS
- Call Forward, Call Hold, Call Transfer
- Dial Plan
- Caller ID
|
Enhanced Features - DHCP Server
- Firmware On-line upgrade
- PSTN Caller ID transmit
- Multiple Language Support
- Supported call divert
- Supported PSTN auto call out to PSTN
- Supported Multi-devices Cooperate
- Mode (Group Mode)
- Supported SMS call out
- Dynamic selection of codec
- Advanced jitter buffer
- Automatic traversal of NAT and firewall
- VLAN / Qos
- Router
- Echo cancellation for Speakerphone
- Comfort noise generation (CNG)
- Voice activity detection (VAD)
- Auto provisioning (requires auto provisioning server)
- On line firmware upgrade
- Multi-language support: English and Chinese
|
Supported Standards - ITU: H.323 V4, H.225, H.235, H.245, H.450
- RFC 1889 RTP/RTCP
- RFC 2327 SDP
- RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
- RFC 2976 SIP INFO Method
- RFC 3261 SIP
- RFC 3264 Offer/Answer model with SDP
- RFC 3515 SIP REFER Method
- RFC 3842 A Message Summary and Message Waiting Indicator
- RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
- RFC 3891 SIP “Replaces” Header
- RFC 3892 SIP Referred-By Mechanism
- draft-ietf-sipping-cc-transfer-04 Session Initiation Protocol Call Control - Transfer
- Codec: G.711 (A/µ law), G.729A/B, G.723.1
- DTMF: RFC 2833, In-band DTMF, SIP INFO
|
Hardware Specifications - Processor: ARM9E 133MHz
- DSP: VPDSP101 95MHz
- Memory: RAM 16MB/ Flash 4MB
- Power: Input AC100V ~ 240V, output 24VDC / 300mA
- Power consumption: 4W maximum
- Network card: 100/10Base-T x2
- LED: Operation and lines light
- RJ11:one 24V feed 48V ring
- Operating temperature: 10°C to 40°C (32°F to 104°F)
- Storage temperature: 0°C to 50°C (32°F to 122°F)
- Working Humidity: 40% ~ 90% Not congealed
- Weight: 95 g (1 lb) (Including AC/DC Adapter)
- Warranty: one year
|
FREE ROAMING
Example:
peer to peer: it is a new function that you can use our VoIP without VoIP provider; what is more, it is free roaming for international call. You just need pay the local call.
Model 1:
Model 2:
Model 3:
Model 4: