DINSTAR DAG1000-2S 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 device 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.
DINSTAR DAG1000-2S Analog Telephone Adaptor 2 FXS
Available on backorder
Model No – DAG1000-1S
- 2 FXS ports, 1 WAN, 1 LAN
- Support SIP, IMS
- T.38 Fax
- Flexible routing & Dial plan
- Interoperable with leading soft switches, IP PBXs and SIP servers
- Support IPv4 and IPv6
- TR069 and SNMP
- Multiple codecs: G.711A/U,G.723.1,G.729A/B, G.726
*Reference to the country telecommunication regulations this product is subject to TRCSL approvals. The product is not available for online purchase. If you are authorized to purchase this product, please submit your company details and obtain a quote or contact our sales team.
Product Availability : On Order Basis, 4-8 Weeks
Physical Interfaces | |
---|---|
LAN | 1 x 10/100Mbps, RJ-45 |
WAN | 1 x 10/100Mbps, RJ-45 |
FXS | 2 x FXS, RJ11 |
Hardware | |
Weight | <= 200g |
Dimensions (W x D x H) | 126 x 76 x 25 mm |
Power Requirements | 100-240VAC, 1A |
Power Consumption | < 5W |
Protocols and Standard | |
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 |
Voice & Fax | G.711A/U law, G.723.1, G.729A/B, G.726 Silence Suppression Comfort Noise Generation(CNG) Voice Activity Detection(VAD) Echo Cancellation(G.168), with up to 128ms Adaptive (Dynamic) Jitter Buffer Hook Flash Programmable Gain Control T.38/Pass-through High speed fax up to 14.4kbps Modem/POS DTMF modes: SIP Info/RFC2833/Inband VLAN 802.1P/802.1Q (voice/data/management VLANs) Layer3 QoS and DiffServ |
VoIP | Protocol: SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,2976,3311 ETC (3GPP TS 24.629, RFC 3515, RFC 3891, RFC 3892) RTP/RTCP, RFC2198, 1889 SIP over TLS RFC4028 Session Timer RFC3266 IPv6 in SDP RFC2806 TEL URI RFC3581 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 |
Supplementary Service | Call Waiting Blind Transfer Attend Transfer Call Forward on Busy Call Forward on No Reply Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-Way Conference Message Waiting Indicator |
Software Features | Hunting Group Web ACL Telnet ACL Action URL PPPoE/IPv4/IPv6 Digitmap Bandwidth Optimization Routing Rules based Prefixes Caller/Called Number Manipulation |
Management | SNMP v1/v2/v3 TR069 Auto Provisioning Web/Telnet Configuration Backup/Restore Firmware Upgrade via Web CDR Syslog NTP Cloud-based Management |
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