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Yeastar D30 DSP Expansion Module
2 × $165.00 -
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Grandstream HT814 Analog FXS IP Gateway - 4 Port + NAT Router
1 × $95.41
Subtotal: $425.41
Dinstar MTG2000 Carrier-Grade Digital E1/T1 Gateway- 20E1/T1
The Dinstar MTG2000 is a carrier-grade intelligent Digital VoIP gateway, scalable from 4 to 20 ports E1/T1. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition.
Description
The Dinstar MTG2000 is a carrier-grade intelligent Digital VoIP gateway, scalable from 4 to 20 ports E1/T1. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition. With highly maintainable, manageable and operable features, it offers a flexible, high-efficient, future-oriented communication network for users.
MTG2000 supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of Trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG2000 is ideal for various applications of large enterprises, call centers, services providers and telecom operators.
Dinstar MTG2000- 4E1/T1 Key Features
- Carrier-grade hardware design, 1+1 power supply
- High-integrated structure, up to 20 E1 ports in 1U size
- Supports Multiple PSTN Protocols: ISDN PRI, R2 MFC, ISDN SS7, etc.
- Supports flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments
- Supports multiple codecs: G.711A/U, G.723.1, G.729A/B and iLBC
- Fully compatible with industry-standard VoIP platforms, including Asterisk, Avaya, Alcatel, Cisco, Huawei, ZTE, etc.
Dinstar MTG2000 Technical Specifications
| brand | Dinstar |
|---|
Physical Interfaces
- E1/T1 Ports 4/8/12/16/20 E1/T1
- DTU Module 4 E1/T1
- Interface Type RJ48(Impedance 120Ω）
- Ethernet Interface GE1: 10/100/1000Â BaseT Adaptive Ethernet GE0: 10/100/1000Â BaseT Adaptive Ethernet
- Serial Port 1* RS232, 115200bps
Voice Capabilities
- Codecs: G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
- Silence Suppression Comfort Noise
- Voice Activity Detection
- Echo Cancellation (G.168),with up to 128ms Adaptive Dynamic Buffer
- Voice, Fax Gain Control FAX:T.38 and Pass-through Support Modem/POS
- DTMF Mode: RFC2833/Signal/In-band Clear Channel/Clear Mode
Environmental
- 1+1 Redundancy Power Supply Power Supply: 100-240VAC, 50-60 Hz Power Consumption:45W
- Operating Temperature:0 ℃ ~ 45 ℃ Storage Temperature: -20 ℃ ~80 ℃ Humidity:10%-90% Non-Condensing Dimensions(W/D/H): 436*300*44.5mm(1U) Unit Weight: 3.8kg
- Compliance: CE, FCC
PSTN
- ISDN PRI 23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
- Signal 7/SS7 ITU-T, ANSI，ITU-CHINA MTP1/MTP2/MTP3, TUP/ISUP
- E1 Frame Type: DF,CRC-4,CRC_ITU
- T1 Frame Type: 4-Frame Multi-frame (F4,FT), 2-Frame Multi-frame (F12, D3/4), Extended Super-frame (F24, ESF) , Remote Switch Mode (F72, SLC96) Line Codes: E1:NRZ,CMI,AMI,HDB3 T1:NRZ,CMI,AMI,B8ZS
- Clock: Local/Remote Clock Source
Maintenance
- Web GUI Configuration Data Backup/Restore PSTN Call Statistics SIP Trunk Call Statistics
- Firmware Upgrade via TFTP/FTP/Web Network Capture
- SNMP v2
- Syslog: Debug, Info, Error, Warning , Notice Call History Records via Syslog, NTP Synchronization, Centralized Management System
Software Features
- Local/Transparent Ring Back Tone Overlapping
- Dialing Rules up to 2000
- PSTN group by E1 port or E1 Timeslot IP Trunk Group Configuration
- Voice Codecs Group
- Caller and Called Number White Lists Caller and Called Number Black Lists Access Rule Lists
- IP Trunk Priority
VoIP Protocol
- SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311
- SIP TLS/SRTP
- RTP/RTCP, RFC2198, 1889
- SIP-T,RFC3372, RFC3204, RFC3398
- SIP Trunk Work Mode : Peer/Access
- SIP/IMS Registration :
- With up to 2000 SIP Accounts NAT: Dynamic NAT, Rport
Call Features
- Flexible Route Methods
- PSTN-PSTN, PSTN-IP, IP-PSTN
- Intelligent Routing Rules Call Routing base on Time
- Call Routing base on Caller/Called Prefixes 256 Route Rules for each Direction
- Caller and Called Number Manipulation
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