Dinstar DAG2500-48S Kuwait

Dinstar DAG2500-48S Kuwait

The Dinstar DAG2500-48S Kuwait is an analog gateway that seamlessly connects traditional phone systems with VoIP. It features 48 FXS ports with a standard R21 interface in a 1U size, allowing users to easily connect analog phones, PBX, and fax devices via a standard voice interface.

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Description

Dinstar DAG2500-48S FXS Gateway Kuwait

The Dinstar DAG2500-48S Kuwait is an analog gateway that seamlessly connects traditional phone systems with VoIP. The Dinstar DAG2500-48S Kuwait features 48 FXS ports with a standard R21 interface in a 1U size, allowing users to easily connect analog phones, PBX, and fax devices via a standard voice interface. This functional gateway supports the standard SIP protocol and is compatible with leading IMS/NGN platforms and SIP-based IP telephony systems. It provides essential telephony services for call centers, small and medium businesses, and multi-location environments requiring VoIP services.

Features :

  • High density gateway, 48 FXS ports in 1U size
  • Open-standard SIP support
  • Support IPv4 and IPv6
  • 3KM Maximum Cabling Length
  • Elastix/Broadsoft Certification
  • Multiple codecs : G.711A/U,G.723.1,G.729A/B, iLBC,AMR
  • Fully compatible with leading IMS/NGN, SIP based IP telephony system

Dinstar DAG2500-48S -Technical Specs :

Cost-Effective VoIP Gateway

  • 48 FXS ports, 2 LAN
  • Support SIP, IMS
  • T.38 Fax
  • Flexible routing & Dial plan
  • Interoperable with leading softswitches, IP PBXs and SIP servers

High Stability and Reliability

  • Embedded Operation System
  • Market-proven hardware design
  • Carrier-grade reliability
  • Main/Secondary SIP server failover
  • TLS/SRTP Security

Easy Management

  • Intuitive Web interface including Quick installation guide
  • Support SNMP &TR-069
  • Automated provisioning
  • Dinstar Cloud Management System
  • Configuration Backup and Restore
  • Debug tools in web interface

General Features

  • IPv6, IPv4
  • Call waiting
  • SIP/IMS
  • Call Transfer (Blind transfer, Attend transfer)
  • SIP v2.0 (RFC3261) based on UDP/TCP/TLS
  • Call Forwarding (Unconditional, No reply)
  • Fax over T.38 and Pass-through
  • Speed Dial
  • QoS : L3 DIFFServ, 802, 1P/Q VLAN Tagging
  • Do Not Disturb(DND)
  • 3-way conference
  • Automatic firmware upgrade and configuration via HTTP/HTTPS
  • Hunting group
  • System logs & CDR
  • Voice Mail
  • Flexible routing & dial plan
  • Pluse Dialing
  • Backup and restore
  • MWI
  • SNMP/TR-069
  • Caller ID
  • TLS/SRTP
  • Music on hold