View source for SMG1000 Series VoIP Gateway

== Quick Links ==
* [[SMG1000 Quick Guide|Quick Guide]]
* [[SMG Series VoIP Gateway FAQ|FAQ]]
* [[SMG1000 Technical Specifications|Technical Specifications]]

== Introduction ==
=== Introduction ===
[[File:Pic-SMG1000-F.png|400px|right|SMG1000 VoIP Gateway]] <br />
 * 2/4/8/16/24/32 FXO/FXS configurable
	
With a simple and economical way to helps legacy telephone, fax machine and PBXs interconnect with IP network,Synway SMG1000 analog media gateway enables call center and multi-branch enterprises to process powerful, versatile and efficient VoIP solutions for unparalleled cost advantage.

SMG1000 Analog Gateways allows for a well-planned, phased migration to an IP network, making the gateways an easy solution for enterprises looking to enhance their legacy PBX equipment with new VoIP access and applications.Connected between a PBX or a Analog and a LAN,the SMG1000 Gateways convert analog PSTN messages into a format suitable for transmission over standard IP networks.<br />

	
*Compatible with general FXO/FXS lines, and a variety of popular PBX manufacturers(Digital PSTN lines compliance would be available)
Protects investment in legacy telecommunications equipment and allows a controlled migration to IP technology

*Developed and tested in Synway PBX lab and optimized for use in an Enterprise environment
Ideally suited for Enterprise Unified Messaging applications

*Support for IP load balancing and IP fault tolerance
Allows the ability for inbound (TDM-to-IP) calls to round-robin between available media servers.

*Supports configuration via serial, telnet, and a web browser including context-sensitive help
Easy to install, configure, debug, and maintain

=== Typical Applications ===
*Centralized VoIP and FoIP application servers, including IP-based voice mail and UMC
*IVR and announcements
*IP PBX
*VoIP extension to branch offices
*Contact centers
	
=== Functional Description ===
Designed for voicemail and unified messaging applications,SMG analog Gateways have a 10/100 Base-T Ethernet connection for connecting legacy PBX to a LAN. The analog loop start functionality supports integration via in-band signaling (DTMF or FSK) or serial protocols. The SMG1000 Gateways provide a simple, cost-effective transition to voice and data convergence for enterprises with PBXs. Connected externally, they offer an IP solution that works with current legacy equipment. They support SIP-based applications as well as T.38 for fax transmissions over IP (FoIP).

{| class="wikitable"
|-
! Key Features !! Values
|-
| Voice over Internet Protocol (VoIP) || Supports SIP per RFC 3261. Uses Real-time Transport Protocol/Real-Time Control Protocol (RTP/RTCP) for delivery of voice over the LAN or WAN
|-
| IP security || Supports HTTPS for web interface
|-
| Enhanced voice processing || Supports a variety of compression algorithms, including G.711 A-law and μ-law, G.729AB
|-
| T.38 Fax over Internet Protocol (FoIP) || Transcode fax from T.30 fax protocol(supporting V.17) to T.38 for transmission over a packet network
|-
| Hot swap || Allows gateway units to be added or removed without affecting other gateway units
|-
| Web server interface || Each gateway unit is delivered with a web server interface, allowing configuration and software upgrades via a web browser
|}	

==== Configurations ====
The SMG1000 Gateways can be used to connect IP telephones to a legacy PBX, integrate network-hosted applications with the PBX, extend the PBX to branch offices, and integrate various voice and call processing capabilities in an enterprise LAN or WAN environment.
Using exclusive PBX network interfaces, the SMG1000 gateway appliances provide exceptional IP to PBX integration capabilities to protect an investment in legacy telecom equipment.

==== Call Routing ====
The SMG1000 Gateways route calls from the switched network to a VoIP destination on the IP network.Conversely, it routes calls from the IP network through a switch port to a destination telephone number on the switched network. The SMG1000 Gateways support the following call routing options:
*TDM to IP or IP to TDM
*IP load balancing
*IP fault tolerance