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Synway SMG1000-D24S Analog VoIP Media Gateway

Synway SMG1000-D24S Analog VoIP Media Gateway enables call centers and multi-branch businesses to process powerful, versatile, and efficient VoIP solutions with unmatched cost advantages. It does this by providing a simple and affordable method for helping legacy telephones, fax machines, and PBXs interconnect with IP networks. The 24 ports FXS VoIP Gateway connects to a PBX, LAN, or WAN and transforms analog PSTN messages into a format appropriate for transmission over common IP networks.
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Description

The Synway SMG1000-D24S is an analog VoIP media gateway that supports FXS interfaces. It is designed to provide seamless connectivity between traditional analog telephony equipment and IP-based voice networks. The gateway is equipped with 24 FXS ports, making it suitable for medium-sized businesses and organizations that require a moderate volume of analog connections. It supports a range of voice codecs, including G.711, G.723.1, G.729A/B, and GSM, ensuring high-quality voice transmission over IP networks. The Synway SMG1000-D24S gateway is easy to install and configure, with a web-based management interface that allows administrators to easily set up and manage the device. It also supports advanced features such as call routing, call forwarding, and caller ID, providing users with a comprehensive set of telephony features. The gateway provides advanced security features, including encryption and authentication, to ensure the privacy and integrity of voice communications. It is a reliable and cost-effective solution for businesses that need to integrate legacy analog telephony equipment with modern IP-based voice networks.

Function

  • Physical Connection Phone Interface: 24 FXS and RJ-11 ports are provided.
    2* RJ-45 10/100 Mbps Ethernet interface For various product variants, Base-T Ethernet, Female RJ-45 1000M LAN/WAN is required.
  • 24 SIP Channels (SMG1000-D24S) and 24 FXS channels for sessions (SMG1000-D24S)
  • Dial Mode: DTMF and Pulse Pulse: 10 and 20PPS for connectivity
  • Phone Number: DTMF/FSK
  • Cable Length Maximum: 5 KM
    Switched Polarity SIP V2.0 OpenVPN TLS/SRTP VoIP Protocols OpenVPN (RFC 3261, 3262, 3264)
    3GPP SDP REFERENCE IMS (RFC 3515) ARP/RARP (RFC 826/903), RTP/RTCP STUN (RFC3489), and
    SNTP (RFC 2030) (RFC 2030) DHCP/PPPoE \sTFTP/HTTP/HTTPS
  • IP Trunks, Call & Routing Port Groups SIP Accounts, both primary and secondary 24 Modification of Inbound/Outbound Routing Numbers Digit load balancing from IP to TDM or IP to IP fault tolerance for IP
  • Voice Functionality iLBC, AMR, G.711A/U legislation, G.723.1, G.729A/B, and G.726
    Comfortable Noise Production (CNG) Cancellation of Echo (G.168) Signal/RFC2833/INBAND is the DTMF mode. Suppression of silence with soothing noise Automatic echo cancellation using G.168
  • Call Progress Analysis (CPA), which includes DTMF detection, Positive Answering Machine Detection (PAMD), Positive Voice Detection, and Fax Tone Detection TDP-IP/IP-TDM call routing is manageable. Limit telemarketing calls with a list of blocked numbers. Voice Activity Monitoring (VAD)
    Adaptive (Dynamic) (Dynamic) Buffer for Jitter Programmable Take Charge Hook Flash

Features

  • Call budgeting based on the allocated amount, minutes, and call count
  • Manageable-based call routing TDP-IP/IP-TDM
  • Restrict unwanted calls with a list of denied numbers
  • Real-time call record send to CDR server
  • Caller ID provisioning

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Amit singh – Acmatel Pvt Ltd

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