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(An ISO-9001 : 2000 Certified Company)
 
 
 
 
     
 
VOIP
 
» ATA - Analog Telephone Adaptor, meets everyone's demand.
 
  The ATA contain four models of gateway products: ATA-One Port FXS, ATA- Two Port FXS, ATA-One Port FXS + One PSTN backup line Port, ATA-One FXS + One FXO with dazzling appearance, ATA series can satisfy all users and meet their different requirements. user can connect with one/two analog phone set to enjoy VoIP application. ATA- One Port FXS + One PSTN backup line Port is one-port analog telephone adapter plus one PSTN backup lifeline, which allows user to dial and receive PSTN/VoIP call in one identical phone set.

 
» Call function
. Call Hold
.
Call Waiting
.
Call Forward
. Caller ID
. Flash
.
Volume Adjustment
.
Speed dial key
. Phone book
. Call Transfer between FXS, FXO and IP port
.
Call Forwarding between FXS, FXO and IP port
. Support T.38 Fax
Technical Specification:
Network and Protocol:
SIP v1(RFC2543), v2(RFC 3261)
IP/ICMP/ARP/RARP/SNTP
TFTP Client/DHCP Client/PPPoE Client
Telnet/HTTP Server
DHCP Server For LAN Port
NAT transversal
Support ToS
 
» Voice feature
. Voice codec A. G.711: 64k bit/s (PCM) B. G.726: 16k/24k/32l/40k bit/s (ADPCM) C. G.729A: 8k bit/s (CS-ACELP) D. G.729B: adds VAD & CNG    to G.729
VAD : Voice activity detection
CNG : Comfortable noise generation
LEC : Line echo canceller
Packet Loss Compensation
• DTMF A. In-Band DTMF, B. Out-Band DTMF C. SIP Info
Tone generation A. Ring Tone B. Ring Back Tone C. Dial Tone D. Busy Tone E. Programming Tone
 
» Application Diagram:
 
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» VoIP Gateway:
 
  VoIP Gateway FXS / FXO are two-port/four-port/Eight-port/twenty four-port/thirty two-port telephone extension to IP network gateway and support H.323 v2/v3/v4 or SIP RFC 3261 protocol. It provides telephone services and T.38 fax over IP network with easy operation and configuration for ITSP / ISP (Internet Telephony Services Provider) and Office / SOHO IP-PBX application.
Benefits
. Easy access to IP from phone set or PBX
. Cost Saving - Telephone call from VPN or public Internet
    ● Follows the existing telephone call dial plan
    ● Easy interface to ADSL/Cable Modem or Leased line equipment
    ● Easy to integrate with all kinds of IP-PBXs
    ● Provision Management via Server
 
» Technical Specifications:
 
Protocol   ITU-T H.323 v2/v3/v4 compliance and SIP RFC3261
Telephony Features   Call Hold, Call Transfer, Call Forward
FAX Support   Automatically FAX detection, T.38 G3 FAX , Group 3 Fax relay at 2.4 - 14.4kbps
Audio Codec Support   G.711A/µ-law, G.723.1, G.729A, G.729 G.729B, G.729AB
Network Interface   10/100Base-T Ethernet RJ-45 port
Voice Quality   VAD (Voice Activity Detection)
CNG (Comfort Noise Generation)
AEC (Acoustic Echo Cancellation)-G.168
Dynamic Jitter Buffer
Caller ID   FSK (Bellcore)/DTMF generation
Network Support   Statically and DHCP for IP address, PPPoE, Behind NAT Router or IP
Configuration/Management   Console port, TELNET and Web Browser
 
» Application Diagram:
 
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» Digital VoIP Gateways
 
The Digital VoIP Gateway cost-effective, converged wire line media gateway. Intelligently packaged in a stackable 1U chassis, it is specially designed to interface between TDM & IP networks in enterprises or small-scale carrier locations. Incorporating Digital Gateway innovative Voice over Packet technology, the Digital VoIP Gateway enables rapid time-to-market and reliable cost-effective deployment of next-generation networks. The Digital Gateways provides superior voice-technology for connecting legacy telephone and PBX systems to IP networks, as well as seamless connection of the IP-PBX to the PSTN. In addition to operating as a pure media gateway, The Digital Gateways are fully interoperable with multiple vendor gateways, soft switches, gatekeepers, proxy servers, IP phones, Session Border Controllers and firewalls.
 
The compact Digital Gateway is extremely scalable and supports multiples of 1, 2, 4, 8 or 16 E1/T1/J1 spans, 4 to 20 BRI ports or 1 to 24 analog ports in various FXO/FXS configurations. It can also supports mixed digital/analog with media processing capabilities such as conferencing, play/record configurations.
 
» Technical Specifications
 
Support Digital Modules 1,2,4,8 or 16, BRI Module, Media Processing Module, Dual Redundant 10/100 Base-TX Ethernet ports via 2 RJ-45 connectors, Voice Codecs -- G.711, G.726, G.723.1, G.729A, GSM-FR, iLBC, EG.711, Independent dynamic vocoder selection per channel, Echo Cancelation -- G.165 and G.168-2002, with 32, 64 or 128 tail length, IP Transport -- VoIP (RTP/RTCP) per IETF RFC 3550 and 3551, Fax and Modem Transport -- T.38 compliant (real time fax), Automatic bypass to PCM,
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» IP Phone
 
IP Phone Solution for cost-savings and flexibility for large organizations or small office / home office users! Incorporate voice and data technologies and eliminate long-distance telephone bills by using state-of-the art VOIP phone on one infrastructure. IP Phone offers exceptional DSP voice quality.
 
Function support and Technical Specifications: call forwarding, call hold/transfer, large number phone book entries, speed dialing, call waiting, caller ID, Duplex speaker phone, stand-alone operation, (no computer needed), 2 x RJ45 Ethernet Ports, Supports G.723.1, G729A/B, G.711 voice codecs, Dynamic IP support (DHCP and PPPoE), Message Waiting indicator, Call History, Volume control, One-Touch memory keys, Session Initiation Protocol (SIP), optional H.323 Protocol, and downloadable software upgrades, and many more..
 
 
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» IP- PBX
 
IP PBX is next generation system for small, medium and large enterprise. It is also designed to operate on a variety of VoIP applications, such as auto-attendant, voice conference, call transfer, call pick up and IP-based communications. With the tiny box, small to medium enterprise or homes can use it to access the Internet and to make VoIP phone calls. Customers could select different suite and optional products to meet their request. To Integrate with VOIP Gateways to provide PSTN connection and IP Phones with flexible and full functionality.
 
» Features of IP-PBX
 
3-way Conference Bridge Call Features - Call Hold / on Hold - Call Transfer (Blind & consultant) - Call Forward (Unconditional / No Answer / Busy / Unavailable) - Call Waiting - Call Pickup - Call Park - Call Camp-On - DND (Do Not Disturb), unlimited no of users registration, "n" no of concurrent con-calls, Call recording, logs, report filtering, extension filtering, user rights, dial plan, etc.
 
» Telephony Features (VoIP)
 
SIPv2 (RFC 3261) - Primary / Secondary register by IP or URI (Number & Text) - Outbound proxy setting for increasing Performance, Productivity, and security - Asterisk based Voice Codecs: - G.711 (a-Law/u-Law), - G.729, -G.723, - G.726, FAX transmission: g.711 pass-through
 
 
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