Dinstar MTG1000 Low Density E1/T1 Digital VoIP Gateway
MTG200 series Digital VoIP Gateways with 1/2 ports E1/T1 simply migrate your legacy PSTN networks (legacy PBX or E1/T1 service providers) , to VoIP network. Only small investment, you can enjoy the real benefits of VoIP, and retain your PSTN connectivity. It is a compact box designed for SMEs and open-source market, fully compatible with Asterisk / Elastix / Trixbox / Freeswitch and mainstream VoIP platform. With support of ISDN PRI / SS7 / R2 MFC, integrating with your legacy PBX or PSTN network is also so easy.
- Efficient concurrent processing, up to 60 channels
- Support flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments
- Support T.38,Pass-through fax, as well as modem and POS machines
- Support multiple coding standards: G.711A/U, G.723.1, G.729A/B and AMR
- High compatibility, interoperable with PBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei, Cisco and ZTE etc.
Physical Interfaces E1/T1 Ports
1 to 2 E1/T1
Interface Type
RJ48(Impedance 120）
Ethernet Interface
GE1: 10/100/1000 Base-T Adaptive Ethernet
GE0: 10/100/1000 Base-T Adaptive Ethernet
Serial Port
1* RS232, 115200bps
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
R2 MFC
China and other 22 variants standard
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
Voice Capabilities Codecs:
G.711a/ law,G.723.1,
G.729A/B, iLBC 13k/15k,AMR
Silence Suppression
Comfort Noise
Voice Activity Detection
Echo Cancellation (G.168), with up to128ms
Adaptive Dynamic Buffer
Voice ,Fax Gain Control
FAX:T.38 and Pass-through
Support Modem/POS
DTMF Mode:
RFC2833/SIP Info/In-band
Clear Channel/Clear Mode
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
VoIP Protocol SIP v2.0 (UDP/TCP),RFC3261
SDP,RTP(RFC2833), RFC3262,
3263,3264,3265,3515,2976,3311
RTP/RTCP, RFC2198, 1889
SIP TLS/SRTP
SIP-T,RFC3372, RFC3204, RFC3398
SIP Trunk Work Mode :Peer/Access
SIP/IMS Registration :
with up to 256 SIP Accounts
NAT: Dynamic NAT, Rport
Software Features Local/Transparent Ring Back Tone
Overlapping Dialing
Dialing Rules，with 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
Environmental 1+1 Redundancy Power Supply
Input 100-240VAC, 50-60 Hz
Power Consumption:15W
Operating Temperature:0 C - 45 C
Storage Temperature: -20 C -80 C
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H): 436*300*44.5mm(1U)
Unit Weight: 2.0kg
Compliance: CE, FCC
Maintenance Web GUI Configuration
Data Backup/Restore
PSTN Call Statistics
SIP Trunk Call Statistics
Firmware Upgrade via TFTP/Web
SNMP v1/v2/v3
Network Capture
Syslog:
Debug, Info, Error, Warning , Notice
Call History Records via Syslog
NTP Synchronization
Centralized Management System
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