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Synway SMG1000-D32O Analog VoIP Media Gateway

The Synway SMG1000-D32O 32Ports FXO Analog VoIP Media Gateway is designed for voicemail and unified messaging applications and includes a 10/100/1000M (optional) Base-T Ethernet connection for connecting legacy PBX to a LAN. The analog loop start functionality allows for integration via in-band signaling (DTMF or FSK), serial protocols, and T.38 for IP-based fax transmissions (FoIP).

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Description

Synway SMG1000-D32O 32Ports FXO Analog VoIP Media Gateway enables call centers and multi-branch enterprises to process powerful, versatile, and efficient VoIP solutions with unparalleled cost advantages by providing a simple and cost-effective way to help legacy telephone, fax machines, and PBXs interconnect with IP networks. The 32-port FXO VoIP Gateway converts analog PSTN messages into a format suitable for transmission over standard IP networks when connected to a PBX, LAN, or WAN. Synway’s 32 ports analog FXO gateway enables call centers and multi-branch enterprises to process powerful, versatile, and efficient VoIP solutions with unparalleled cost advantages by providing a simple and cost-effective way to help legacy telephone, fax machines, and PBXs interconnect with IP networks.

The 32-port FXO VoIP Gateway converts analog PSTN messages into a format suitable for transmission over standard IP networks when connected to a PBX, LAN, or WAN. The 32 Ports Analog VoIP Gateway SMG1000-D32O is designed for voicemail and unified messaging applications and includes a 10/100/1000M (optional) Base-T Ethernet connection for connecting legacy PBX to a LAN. The analog loop start functionality allows for integration via in-band signaling (DTMF or FSK), serial protocols, and T.38 for IP fax transmissions (FoIP).

Features

  • Easy integration with existing telephony interfaces
  • Open-standard SIP support and registration to multiple SIP proxy servers
  • Make and receive IP calls from analog extensions
  • 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

Benefits

  • Voice optimization to ensure better user experiences
  • Enhanced call routing capability with high voice quality
  • Easy to install, configure, and maintain
  • Support IPv4 and IPv6 international networks
  • Data/voice/management VLAN and more
  • Built-in firewall and access rules
  • Support for SNMP/TR069/Auto-Provision
  • Cloud-based management and bandwidth optimization
  • Support for SIP, MGCP, and other customizable protocols
  • Primary/Backup SIP Servers

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