শনিবার, ২২ সেপ্টেম্বর, ২০১২

Yeastar - NeoGate - voip-info.org

Designed for maximum cost reduction

NeoGate TG200 is a modular VoIP GSM/UMTS gateway with 1/2 channels providing GSM /UMTS network connectivity for soft switches, and IP-PBXs. It supports two-way communication: VoIP to GSM/UMTS and GSM/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM network.

Benefits

1) Cost Savings ? Cost-Savings on phone calls between mobiles or to PSTN.
2) Back up ? Should the landline network go down, GSM can be used as a cost-effective backup.
3) Easy to install ? IP device with Web based management interface.
4) Easy to integrate.

Specification:

Number of GSM channels (Max): 2
Network type
UMTS: 900/2100MHz; 850/2100MHz; 850/1900MHz;
GSM: 850/900/1800/1900MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

LAN: 1 (10/100Mbps)
Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 200x140x35mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

SIP proxy Registrar for IP phones included
Incoming call routing
Outgoing call routing
SMS sending and receiving (WEB interface)
Call Back
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
Simple web based configuration
Easy to integrate / Easy to install

Smooth communication between BRI and VoIP network

NeoGate TB400 is a compact reliable standalone VoIP BRI gateway (BRI-VoIP/VoIP-BRI) offering the company using ISDN BRI lines an easy, cost-effective and flexible integration into any VoIP system or enabling any IP PBX to be connected to the public ISDN network at an affordable price. It could either provide VoIP access for your legacy PABX or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

Specification:

BRI Port: 4 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 200x140x35mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install

100% software based, Generate up to 30 Skype trunks!

Nowadays, Skype is very popular and you may found many customers are Skype users. Let your customers who are used to Skype to contact with you quickly and conveniently is becoming the main job of your Asterisk/IPPBX system. SiSkyEE is the best solution for you to connect PBX to the Skype world.

Functions

Website Click-to-Call: Receive calls from website and SkypeIn number with multi-trunk.
Interoffice Trunking: Builds enterprise branch communications network through Skype with optimal design .
Skype Trunking:
  • Skype Incoming : Receive Skype calls from customers who are using Skype.
  • Outgoing to Skype : Make free calls to numerous Skype users by office phone.
  • SkypeOut: Provide trunks of making landline/mobile calls.
  • Remote Extension: Use Skype as Fixed Remote Extension.
  • SiSky include a SIP Server, it can work as Asterisk'/IPPBX's SIP Trunk or SIP Extension.
  • Remote monitoring and managing by Web.
  • Independent Phonebook Utility includes public & private phonebook.
  • Optional Multi-User mode feature allows every user to create his own private contacts.
  • Play the auto attendant for Skype incoming call, and forward it to extension or Skype.
  • Call Log & Call Statistics.
  • Database sharable among cascade connection of SiSky Servers.
  • Backup and restore functions of database.

Features

  • Supports 30 Skype trunks (concurrent calls) on one computer.
  • Builds enterprise branch communications network through Skype with optimal design.
  • Company Skype ID, add the effective voice trunk to Internet.
  • Automatically finds idle trunk to transfer Skype Incoming calls.
  • Automatically finds idle trunk to make Skype Outgoing calls.
  • Delivers Skype functionality into enterprise extension system.
  • Enable customized Speed-dial or PSTN matchable dialing plan.
  • Sets a dedicated phone extension to ring for Skype incoming call.
  • Multi-User Mode allows users to create and manage their own contacts.
  • 'Utility' allows every user to export his personal Skype contacts into his private phonebook.
  • Noise reduction, echo cancellation and compensation for losing packet techniques ensure the excellent voice quality.
  • Load Balancing: system will distribute the flow rate equally among channels.


Try before buy, you run no risk in purchasing SiSkyEE. Here you can download the software directly, and try to use it to see if its performance could meet your requirement.

Source: http://www.voip-info.org/wiki/view/Yeastar+-+NeoGate

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