Product Details

Specifications

unlock voip adapter
Have encryption built-in;
Support VPN;
Have router fuction;
Can be customized.

1 FXS & 1 FXO VOIP SIP GATEWAY

The 1-FXS 1-FXO VoIP Gateway HTM-112 is the performance Gateway of FXO plus FXS. HTM-112 built-in H.323 and SIP protocols. Highly reliable line detection prevents the line hanging death in the largest tent. It can forward outside Caller ID Number under the SIP protocol, which is an important function of PBX application. set connected to the FXS port can also access the PSTN line for traditional telephone service. The HTM-112 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.

Key Features

Static IP support
Switch mold support
support DHCP,PPPOE
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

Two RJ-11 FXS port for traditional phone set or PBX's trunk line
Two RJ-11 port for connection to PSTN line(bypass function only)
LEDs for Power, Ready, Status, WAN, PC, FXS
Call Forward, Call Hold, Call Transfer
Dial Plan
Caller ID
3 times re-password
Billing support
Keyboard setting
IP voice enrollment

Enhanced Features

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