DINSTAR DAG2500 is a functional analog VoIP gateway offering seamless connectivity between VoIP and traditional phone system. It’s a highly-integrated gateway which provides up to 72 FXS ports with standard R21 interface in 1U size. Users can easily connect their analog phones, fax machines and PBX systems via standard voice interface. DAG2500 supports the standard SIP protocol and it’s compatible with leading IMS/NGN platform and SIP-based IP Phone system. It is ideally suited for small and medium businesses, call centers and multi-location environments that need VoIP services.
DINSTAR DAG2500-72S 72 FXS Analog VoIP Gateway
Available on backorder
Model No – DAG2500-72S
- 72 FXS ports, 2 LAN
- Support SIP, IMS
- T.38 Fax
- Flexible routing & Dial plan
- Interoperable with leading softswitches, IP PBXs and SIP servers
- Support IPv4 and IPv6
- 5KM Maximum Cabling Length
- Multiple codecs: G.711A/U,G.723.1,G.729A/B, iLBC.
*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 | 2 x 10/100Mbps |
FXS | 72 x FXS, RJ21 |
Console | 1 x RS232, 115200bps |
Hardware | |
Unit Weigh | 4 kg |
Dimensions | 440 x 280 x 44 mm |
Power Requirements | 100-240VAC, 50-60Hz |
Power Consumption | 75W |
Protocols and Standard | |
FXS | Connector: RJ21 Dial Mode: DTMF and Pulse Pulse: 10 and 20 PPS Caller ID: DTMF/FSK CLI Presentation Max Cable Length: 5KM Reversed Polarity Programmable Call Progress Tone |
Voice & Fax | G.711A/U law, G.723.1, G.729A/B, G.726,iLBC 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 Modem/POS DTMF mode: Signal/RFC2833/INBAND VLAN 802.1P/802.1Q Layer3 QoS and DiffServ |
VoIP | Protocol: SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311 SIP TLS RTP/RTCP, RFC2198, 1889 RFC4028 Session Timer RFC3266 IPv6 in SDP RFC2806 TEL URI RFC3581 NAT,rport Outbound Proxy DNS SRV/ A Query/NATPR Query SIP Trunk Early Media/Early Answer NAT:STUN, Static/Dynamic NAT |
Supplement Service | Call Waiting Blind Transfer Attend Transfer Call Pickup 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 TR069 Auto Provisioning Web/Telnet Configuration Backup/Restore Firmware Upgrade via Web CDR Syslog NTP Cloud-based Management |
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