Gold Member Since 2010
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Shenzhen DBL Technology Limited

SIP, VoIP Gateway, Internet Access Gateway manufacturer / supplier in China, offering FXS Port VoIP Gateway/H. 323&SIP/Unlimited Global Call (HT-912T), GoIP 8 8 Channel VoIP GSM Gateway/ 8 SIM Card Wireless VoIP Terminal Support SMS Server, 8 Port/Channel GSM VoIP Gateway with SIP&H. 323 (GoIP8) and so on.

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Supplier Homepage Product VoIP ATAs(FXS) FXS Port VoIP Gateway/H. 323&SIP/Unlimited Global Call (HT-912T)

FXS Port VoIP Gateway/H. 323&SIP/Unlimited Global Call (HT-912T)

FOB Price: US $26 / Piece
Min. Order: 1 Piece
Min. Order FOB Price
1 Piece US $26/ Piece
Get Latest Price
Port: Shenzhen, China
Production Capacity: 10000PCS/Month
Payment Terms: T/T, Western Union, Paypal
Basic Info
  • Model NO.: HT-912T
  • Protocol: H.323&SIP
  • Number of Channels: 1
  • Color: Black
  • Trademark: DBL
  • Origin: China
  • Type: Gateway
  • Support: VPN
  • Wire or Not: Wire
  • Warranty: one year
  • Transport Package: 14*17*6.5cm
Product Description
What is the function of VoIP ATA(FXS)?
VoIP ATA (Analog Terminal Adapter) is a telephone extension to the IP network. It offers a traditional telephone line (PSTN) interface to an analog telephone, PBX line extension, or a fax machine. Its WAN port interface allows access to the IP network in order to offer voice and fax services.

Key Features
Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Peer-to-Peer IP Calls
Two 10/100 Ethernet for WAN / LAN connections
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
Four RJ-11 FXS ports for traditional phone sets or PBX's trunk lines
LEDs for Power, Ready, Status, WAN, PC, FXS ports
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

Physical and Environmental
Operating temperature: 10° C to 40° C (50° F to 104° F)

Storage temperature: 0° C to 50° C (32° F to 122° F)

Weight: 0.10 kg (4.2 lb) (Including AC/DC Adapter)

Size: 70mm (W) x 101mm (L) x 25mm (H) ( " x " x ")

Power: 12 Vdc 500 mA (AC/DC adapter included). Classification 0 under IEEE 802.3af on CN2x4

Warranty: One year
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