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DAG1000-8O FXO GATEWAY

FIBERME Analog FXO Gateway FAG4104
Yeastar TA810 (8 Ports)
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Description

The DAG1000-8O FXO Analog VoIP Gateway is a type of access gateway offering seamless connectivity between IP-based telephony networks and legacy telephones (POTS) and PBX systems. The analog gateway has 8 FXO ports and is used to connect to analog PBX or the PSTN lines of telecom carriers. With the standard SIP protocol, it’s compatible with leading IMS/NGN platforms and SIP-based IP Phone systems. It provides low-cost and easy-to-use VoIP solutions for small and medium businesses, call centers, SOHO, remote offices as well as enterprises with multiple branches.

Dinstar DAG1000-8O Key Features

  • Cost effective gateway with 4/8 FXO ports
  • Fax over IP (T.38 and Pass-Through)
  • Support IPv4 and IPv6•TR069 and SNMP
  • Multiple codecs: G.711A/U,G.723.1,G.729A/B etc.
  • Fully compatible with leading IMS/NGN, SIP based IP telephony system
Dinstar DAG1000-8O – Technical Specifications

Key Features

  • 8 FXO ports, 1 WAN, 3 LAN
  • Support SIP, IMS
  • T.38 Fax
  • Flexible routing & Dial plan
  • Interoperable with leading softswitches, IP PBXs and SIP servers
  • Embedded Operation System
  • Market-proven hardware design
  • Carrier-grade reliability
  • Main/Secondary SIP server failover
  • TLS/SRTP Security

Voice Processing

  • Voice Codec: G.711A/U law, G.723.1, G.729A/B
  • Voice Packet Capabilities: CNG, VAD, G.168, AJB
  • Programmable Gain Control
  • DTMF mode: Signal/RFC2833/INBAND

Call Features

  • Calling Type: Voice/ T.38/Pass-through/Modem FAX mode, Up to 14.4kbps
  • IVR Voice Prompt: Private service IVR
  • Telephony Features: Call waiting, Call transfer(Blind  transfer, Attend Transfer , Half  blind transfer), Quick pick, Call forwarding, Hotline, Call hold, DND,  Voice mail, 3-way conference,Direct IP address call

Network

  • Network Protocols: IP,TCP, UDP,TFTP, FTP, ARP,RARP, Ping, NTP, SNTP, Http, DNS, Trancert
  • NAT Traversal: STUN, Ping, Tracert

Protocol and Standard

  • Protocol: SIP V2.0 (RFC 3261,3262,3264), RTP/RTCP
  • Standard: Bellcore Type 1 &2, ETSI, BT, NTT and DTMF based CID, configurable call progress tone

Configuration Management

  • Auto Provisioning: TFTP, HTTP, Auto configure/upgrade, Syslog and CDR, SNMP
  • Web GUI: Configuration, firmware upgrade, TCP/UDP/Current RTP/History calls statistics

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