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SHENZHEN DINSTAR CO., LTD.

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Products: VoIP Phone , GSM Gateway , VoIP Gateway , IP Phone , FXS VoIP Gateway , FXO VoIP Gateway , E1 Gateway , VoIP , Wireless , Asterisk

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ippbx for call center selution Carrier and enterprise-class devices16 port FXS sip VOIP gateway, goi

OverviewQuick DetailsPlace of Origin:Guangdong, China (Mainland)Brand Name:DinstarModel Number:DAG2000-16SType:VoIP GatewayProduct Type:SIP

$100 /


Stock:	10000 s
Quantity:	-+ Min Order: piece

Product Details

Transaction Comments

Overview

Quick Details

Place of Origin:: Guangdong, China (Mainland)

Brand Name:: Dinstar

Model Number:: DAG2000-16S

Type:: VoIP Gateway

Product Type:: SIP VoIP access gateway

Number of interface:: 1*WAN 10/100 BASE-TX (RJ-45)

Number of Interfaces:: 16 * FXS

Transmission distance:: 1km

Rate:: 115200bps

Connector type:: DB-9 connector

Protocols compliant:: SIP V1.0/2.0 and RFC3261

Voice over Packet:: VAD, CNG, DJB and PLC

Voice Codec:: G.723.1, G.729A/B and PCMA/PCMU

Supply Ability

Supply Ability:: 10000 Piece/Pieces per Month

Packaging & Delivery

Port: shenzhen/ HK

DAG2000-16S is a multi-functional analog gateway offering seamless connectivity between IP-based telephony networks and legacy telephones (POTS), fax machines and PBX systems. With the standard SIP protocol, it's compatible with leading IMS/NGN platforms and SIP-based IP telephony systems. The device offers 16 FXS ports, fax over IP and flexible dial plans. It is ideally suited for small and medium businesses, call centers and multi-location environments that need VoIP services.

ippbx for call center selution Carrier and enterprise-class devices16 port FXS sip VOIP gateway, goip sip analog gateway

Physical Interface
1. Telephone Interface:
2. 16* FXS, RJ11
3. Network Interface:
4. 4* LAN, 10/100Mbps, RJ45
5. Console: 1*RS232, 115200bps
FXS
1. Connection Interface: RJ11
2. Caller ID: DTMF/ FSK
3. Support Reverse Polarity
4. Dial mode: DTMF and Pulse
5. Pulse: 10 and 20 PPS
6. Maximum wiring length: 3000m
7. Programmable Call Progress Tone
Supplementary Business
1. Pick Up on Group
2. 3-Way Conference
3. Busy Call Forwarding
4. No answer Call Forwarding
5. Unconditional Call Forwarding
6. Blind Transfer, Attend Transfer
7. Support Warm/Immediate Hotline
8. Message Waiting Indicator (Voicemail)
9. Call Waiting, Call Hold, Do-Not-Disturb
Software Features
1. Action URL
2. Hunting Group
3. Web access control rules
4. Telnet access control rules
5. PPPoE/IPv4/IPv6(Bridge mode)
6. Routing Rules based on Prefixes
7. The number of graphs(Digitmap)
8. Caller/Called Number Manipulation

Voice Processing
1. VLAN 802.1P/802.1Q
2. Layer3 QoS and DiffServ
3. Coder: G.711A/U law, G.729A/B,
4. G.723.1, G.726, iLBC and AMR
5. Programmable Gain Control
6. Voice Activity Detection (VAD)
7. Adaptive (Dynamic) Jitter Buffer
8. Hook Flash, Silence Suppression
9. T.38/Pass-through, Modem/POS
10. Comfort Noise Generation (CNG)
11. Echo Cancellation (G.168), with maximum 128ms
12. DTMF mode: Signal/RFC2833/Inband
VoIP Protocol
1. RFC3266 IPv6 in SDP
2. RFC4028 Session Timer
3. RTP/RTCP, RFC2198, 1889
4. Early Media/ Early Answer
5. Primary/Backup SIP server
6. SIP Trunks, Outbound Proxy
7. STUN, Static/ Dynamic NAT, DTR
8. DNS SRV/ A Query/ NATPR Query
9. RFC2806 TEL URI, RFC3581 NAT, rport
10. Protocol: SIP v2.0 (UDP/TCP), RFC3261 SDP, RTP(RFC2833), RFC3262, 3263, 3264, 3265, 3515, 2976, 3311
Maintenance
1. Web/Telnet
2. Ping/Tracert Test
3. Network Capture
4. Auto Provisioning
5. IVR local Maintenance
6. SNMP v1/v2/v3, TR069
7. DMCloud Management
8. Line Diagnostics (GR909)
9. Firmware Upgrade via Web
10. Configuration Backup/Restore
11. Syslog, NTP/Summer Time, CDR
Environmental
1. Power Adapter:
2. 100-240VAC, 50-60 Hz
3. Power Consumption: 30W(Typical)
4. Operating Temperature: 0 ℃ ~ 45 ℃
5. Storage Temperature: -20 ℃ ~80 ℃
6. Humidity: 10%-90% Non-Condensing
7. Dimensions(W/D/H): 440*230*44mm
8. Unit Weight: 2.7kg
9. Compliance: CE, FCC

1/2FXS/4FXS/4FXO,8FXS/8FXO,16FXS/16FXO multiple interfaces

Standard SIP V2.0, MGCP V1.0 protocols

Standard G.711A/U,G.723.1,G.729A/B,G.168,PCM A/U code

Call forwarding, Call hold, Caller ID, DND standard voice service

Busy tone, Power-down protection , Fault identification

Many users sharing ADSL lines

DAG series is a versatile IP-based voice and fax gateway, it is highly integrated router, NAT, DHCP server, phone gateway functionality. Market-leading voice quality, rich functionality, and lightweight, compact design. DAG series is fully compatible with SIP industry standard, can be achieved with many other SIP compliant devices and software interoperability. In addition, it supports including G.711A / U, G.723.1 and G.729AB voice codec.

Carrier and enterprise-class devices, both low-cost VoIP service while preserving the traditional PSTN network. Especially for new VoIP and traditional PSTN operators to seize market operators.

DAG series can be with the mainstream SIP proxy server(such as Asterisk, SIP registration server, etc. ) to implement docking. In this kind of environment, SIP server processing SIP registration and call control, DAG series access gateway can realize flow media conversion between the IP and PSTN.