Dinstar MTG200 Cost-effective VoIP Trunk 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.
- Up to 60 simultaneous calls with 2* E1/T1 ports
- Support echo cancellation, DJB, CNG, VAD and QoS
- Use of existing E1/T1 ISDN PRI interface of the PBX
- Use of existing VoIP interface of the IP-PBX
- Maintain existing dialing habits and business communication patterns
- Built-in Web-based management, SNMP, command line interface (CLI)
- Standard ISDN compliant and Interoperable with a wide range of IP-PBXs
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
1x RS232, 115200bps
PSTN ISDN PRI
23B+D(T1),30B+D(E1),NT or TE
ITU-T Q.921, ITU-T Q.931, Q.Sig
SS7 (optional)
ITU-T, ANSI，ITU-CHINA
MTP1/MTP2/MTP3, TUP/ISUP
R2 MFC (optional)
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 Default codec:G.711a/μ law, AMR
G.723.1, G.729AB, iLBC
Silence Suppression
Comfort Noise
Voice Activity Detection
Echo Cancellation (G.168), up to 128ms
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
VLAN 802.1p/q
60 Concurrent Calls
Software Features Local/Transparent Ring Back Tone
Overlapping Dialing
Dialing Rules，with up to 2000
Voice Codecs Group
Access Rule Lists
100 SIP Trunks
Route direction:
PSTN-IP, IP-PSTN, PSTN-PSTN,
IP-IP
VoIP Protocol SIP v2.0 (UDP/TCP),RFC3261
SDP,RTP(RFC2833), RFC3262,
3263,3264,3265,3515,2976,3311
RTP/RTCP, RFC2198, 1889
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
Maintenance Software Features
Web GUI Configuration, HTTP/HTTPS
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
Device Events to Email
Centralized Management System
Environmental Power Adapter: 100-240VACDC12V1A
Power Consumption:15W
Operating Temperature:0 C - 45 C
Storage Temperature: -20 C -80 C
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H): 225x150x38mm
Unit Weight: 0.8kg
Compliance: CE, FCC
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