Specifications
1)one fsx+one fxo
2)SIP and H.323
3)NAT transversal and route function
4)Vlan and QoS
5)high quality,good price,
OverviewHTM-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. |
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Key Features- 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
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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
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
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Application:
Saving Cost on International and Long Distance Calls,when you are in the other countries& you have this gateway at your Local office,you could dial & receive phone call from your customers as you are there without Long Distance fee.
Very excellent solution for internal company with branch offices oversea.
SKYLINE International Communication Co., ltd is a high-tech enterprise integrated with scientific research, development, production and sales. The company has been engaged in the research, development and production of communication products since its establishment. Our products are listed as follows.
FXS series:HT-912(one fxs port),HT-922(two fxs ports),HT-842R(four fxs ports).HT-882(eight fxs ports)
FXS+PSTN series:HT-812P(one fxs port with a PSTN live line),HT-822P(two fxs ports with a PSTN live line),HT-522(two fxs ports with two PSTN live lines),HT-544(four fxs ports with four PSTN live lines)
FXO series:HT-322(two fxo ports),HT-342(four fxo ports)
FXO+FXS series:HT-112(one fxo port and one fxs port),HT-222(two fxo ports and two fxs ports),HT-442(four fxo ports and four fxs ports)
GSM&CDMA series:GoIP 1 (one channel), GoIP 4 (four channels), GoIP 8 (eight channels), CoIP 4 (four channels)
RoIP series: RoIP302, RoIP302M, RoIP102
IP Phone: EP-636 (one or two lines), EP-8201 (four lines)
Recorder Box:TYH8200 (one line), TYH8201 (two lines), TYH636 (eight lines)
If you have any doubt or want to know further information,please feel free to contact me.