US$ 43.08 - 43.09 MOQ: 1 Piece
Key Specifications
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Place of Origin:
Guangdong, China
Brand Name:
SKYLINE
Model Number:
HT-912, HT-912
Payment & Shipping
Payment Methods:
Port of Shipment:
SHENZHEN
Delivery Detail:
Delivery time depends on order quantity.
Place of Origin Guangdong, China
Brand Name SKYLINE
Model Number HT-912, HT-912
Type VoIP Phone

Specifications

sip gateway :
1 support ITU H.323V4 and IETF SIPV2.
2 high quality speech
3 support VLAN and QoS
4 NAT Transversal

1 channel FXS Voip sip gateway of free roaming

Product Introduction:

The HT-912 VoIP is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It comes with one FXS port to interface with a traditional analog phone set or a PBX trunk line for VoIP communications.  It bundles with lots of features to meet the demand in various network environment.  It is an ideal low cost VoIP solution for travelers and SOHO users.

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)
  • 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:

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Model 2:

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Model 3:

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Model 4:

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Verified Business License
Business Type
Trading Company
Year Established
2007
Factory Size
Below 1,000 square meters
Total Employees
201 - 300 People