Description
Dinstar DAG1000-8S-GE 8 FXS Analog VoIP Gateway Oman
The Dinstar DAG1000-8S-GE Oman is a powerful, versatile, and highly reliable multi-functional analog gateway engineered to provide seamless integration between modern IP-based telephony systems and a wide range of traditional analog communication devices, including legacy POTS phones, fax machines, and existing PBX infrastructures. Featuring 8 FXS RJ11 ports combined with dual Gigabit Ethernet interfaces, it ensures outstanding voice quality, stable performance, and fast data transmission across both LAN and WAN environments, making it suitable for demanding communication setups. Its support for advanced Fax-over-IP technologies such as T.38 and Pass-through, paired with flexible and fully customizable dial plan configurations, enables the device to adapt easily to diverse usage scenarios and evolving business communication requirements. Ideal for small and medium-sized businesses, call centers, remote branch offices, and expanding enterprise networks, the Dinstar DAG1000-8S-GE not only streamlines VoIP migration but also improves system administration, enhances daily operational efficiency, and delivers a cost-effective, forward-looking solution that effectively connects traditional telephony systems with modern IP communication networks, ensuring smooth, scalable, and future-ready performance.
Features
Cost-Effective VoIP Gateway
- 8 FXS ports, 1 WAN, 1 LAN
- Support SIP, IMS
- 38 Fax
- Flexible routing & Dial plan
- Interoperable with leading soft switches, 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
Technical Highlights
- 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
- 5 km long haul
- TLS/SRTP
- Music on hold












































