Description
Dinstar DAG1000-2S analog telephone adapter Ghana
The Dinstar DAG1000-2S Ghana is a multi-functional analog telephone adapter which offers seamless connectivity between IP Telephony networks and legacy telephones (POTS), fax machines and PBX systems. The Dinstar DAG1000-2S analog telephone adapter provides 2 FXS ports, fax over IP and a built-in high-speed NAT router. These powerful features and good voice quality make the device ideal for various application environments such as SOHO and small enterprises and for personal use. Moreover, with automatic provisioning and centralized management system, the device is easy to maintain and deploy.
Key Features
Cost Effective VoIP Management
- 1/2 FXS ports, 1 WAN, 1 LAN
- Support SIP, IMS
- 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
Technical Specifications
Physical Interfaces
- Telephone Port
- DAG1000-2S: 2 FXS ports (RJ11)
Ethernet Interfaces
- DAG1000-2S:
- 1 WAN, 10/100Mbps (RJ45)
- 1 LAN, 10/100Mbps (RJ45)
Voice Capabilities & Fax
- Codecs: G.711a/μ law, G.723.1, G.729A/B, G.726
- Silence Suppression
- Comfort Noise Generator(CNG)
- Voice Activity Detection(VAD)
- Echo Cancellation: G.168 with up to 128ms
- Adaptive (Dynamic) Jitter Buffer
- Hook Flash
- Adjustable Gain Control
- Programmable Gain Control
- FAX: T.38 and Pass-through
- Modem/POS
- DTMF: SIP Info/RFC2833/Inband
- VLAN 802.1P/802.1Q
- Layer 3 QoS and DiffServ
FXS
- FXS Connector: RJ11
- Dial Mode: DTMF and Pulse
- Pulse: 10 and 20 PPS
- Caller ID: DTMF/FSK CLI Presentation
- Max Cable Length: 3km
- Reversed Polarity
- Programmable Call Progress Tone
VoIP
- Protocols: SIP v2.0 (UDP/TCP), RFC3261, SDP,
RTP(RFC2833), RFC3262, RFC3263,RFC3264,
RFC3265, RFC3515, RFC2976, RFC3311 - RTP/RTCP, RFC2198, RFC1889
- SIP over TLS
- RFC4028 Session Timer
- RFC3266 IPv6 in SDP URI
- RFC 3581 NAT.rport
- Primary/Backup SIP Server
- Outbound Proxy
- DNS SRV/A Query/NATPR Query
- SIP Trunk
- Early Media/Early Answer\
- NAT: STUN, Static/Dynamic NAT
Software Features
- Hunting Group
- Web ACL
- Telnet ACL
- Action URL
- PPPoE/IPv4/IPv6
- Digitmap
- Bandwidth Optimization
- Routing Rules based on Prefixes
- Caller/Called Number Manipulation
Supplementary Services
- Call Waiting and Call Holding
- Call Forwarding (Unconditional/Busy/No Reply)
- Call Transfer (Blind & Attended)
- Warm/Immediately Hotline
- Do-not-disturb
- Three Parties Conversation (3-way Conference)
- Message Waiting Indicator
Environmental
- Power Supply: 12V DC, 1A
- Power Consumption: <5W
- Operating Temperature: 0 ℃~ 45 ℃
- Storage Temperature: -20 ℃ ~80 ℃
- Humidity: 10%-90% (Non-Condensing)
- Dimensions:126×76×25mm(W/D/H)
- Unit Weight:<=0.2kg
- Compliance: UL
Maintenance
- SNMP V1/V2/V3
- TR069
- Auto Provisioning (HTTP/FTP/TFTP)
- Web/Telnet
- Configuration Backup/Restore
- Firmware Upgrade via Web
- CDR
- Syslog
- Ping, Tracert Test
Network Capture
- Outward Test (GR909 Standard)
- NTP/Daylight Saving Time
- IVR Local Maintenance
- Cloud-based Management