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Dinstar VoIP Gateway DAG1000-1S1O
Hybrid Analog VoIP Gateway, 1 FXS & 1 FXO, SIP, SNMP, Failover Lifeline, Voice/Fax Support
Dinstar DAG1000-1S1O is a hybrid analog VoIP gateway with 1 FXS and 1 FXO port, designed for small businesses, remote offices, and branch deployments. It offers seamless VoIP integration with traditional phone systems through failover lifeline support for power and network outages. Fully compatible with SIP-based IPPBX platforms, the gateway supports voice and fax services, SNMP, TR069, and advanced codec handling. With high voice quality, programmable routing, echo cancellation, and easy provisioning, the DAG1000-1S1O is an efficient and cost-effective analog-to-IP solution.
Key Features:
- 1 FXS and 1 FXO analog ports
- Failover lifeline for power and network failure
- SIP support with SIP TLS and SRTP
- G.711, G.729, G.723, G.726, AMR codec support
- Voice and fax over IP with T.38 and pass-through
- SNMP, TR069, web and cloud-based management
- Echo cancellation up to 128ms, jitter buffer, VAD, CNG
- Compatible with major IPPBX and softswitch platforms
- Caller ID, polarity reversal, call detection, digit map
- 1 LAN and 1 WAN Ethernet port
- Auto provisioning, firmware upgrade, remote config
- Compact design with 10W power consumption
Product Specifications:
- Manufacturer: Dinstar
- Model: DAG1000-1S1O
- Phone Interfaces: 1 FXS, 1 FXO (RJ11)
- Ethernet Interfaces: 1 LAN, 1 WAN (RJ45, 10/100Mbps)
- Power Supply: Input 100–240VAC; Output 12VDC, 2A
- Power Consumption: 10W
- Operating Temperature: 0°C to 45°C
- Storage Temperature: -20°C to 80°C
- Humidity: 10%–90% (non-condensing)
- Dimensions: 126 × 76 × 25 mm
- Weight: 0.2 kg
- Compliance: CE, FCC
- Voice Codecs: G.711A/U, G.723.1, G.729A/B, AMR
- Fax Support: T.38, pass-through
- DTMF Mode: Signal, RFC2833, Inband
- Echo Cancellation: G.168, up to 128ms
- Network Protocols: IPv4, TCP, UDP, PPPoE, DHCP, HTTP/S, SNMP, OpenVPN
- Management: Web, Telnet, SNMP v1/v2/v3, TR069, cloud
- Features: Auto provisioning, CDR, syslog, IVR, digitmap, ACL, NAT traversal, VLAN, QoS
- Call Control: SIP v2.0, SIP trunk, early media, STUN, proxy
- Caller ID: DTMF, FSK
- Call Detection: Bellcore Type 1/2, ETSI
- Routing: Prefix-based, speed dial, number manipulation
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