VoIP Isdn R2 Gateway Mtg200 to SIP

Product Details
Customization: Available
Type: Gateway
Protocol: TCP/IP
Manufacturer/Factory & Trading Company
Gold Member Since 2011

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Number of Employees
152
Year of Establishment
2011-02-23
  • VoIP Isdn R2 Gateway Mtg200 to SIP
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Basic Info.

Model NO.
MTG200
Support
SIP
Number of Channels
4
Wire or Not
Wire
Color
Black
E1 / T1
1/2/4
Concurrent Calls
up to 120
SIP Trunks
32
Tls/Srtp
Yes
Transport Package
Gift Box
Specification
CE, ISO90001
Trademark
DINSTAR
Origin
China
HS Code
851762
Production Capacity
100000PCS/Year

Product Description

Desktop Digital VoIP E1/T1 Gateway MTG200
Dinstar MTG200 Trunk Gateway enables direct routing of calls between the fixed line ISDN and the cost-effective IP networks for capitalizing on low cost VoIP telephony. By integrating MTG200 Trunk Gateway with existing PBX / PABX telephone systems, businesses of all sizes can benefit from low-cost Voice over IP calls and achieve substantial cost savings without the high upfront costs associated with communications infrastructure changes related with the migration to VoIP.
MTG200 offering PBX/IP-PBX users a powerful telephony cost saving device and advanced flexible routing capabilities between ISDN PRI to SIP, and also can be situated between the PBX and the PSTN, saving the cost of an additional PBX E1/T1 port. It is ideally fit for various access networks of SMEs, call centers, telecom operators and large-scale enterprises.


 


Key Features
Up to 120 simultaneous calls with 4* 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 4 E1/T1
Interface Type
RJ48(Impedance 120Ω)
Ethernet Interface
GE1: 10/100/1000 Base-T Adaptive Ethernet
GE0: 10/100/1000 Base-T Adaptive Ethernet
Serial Port
1* RS232, 115200bps 

Voice Capabilities
Default codec:G.711a/μ law, AMR
G.723.1, G.729AB, iLBC (License)
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
 
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

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

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
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
Centralized Management System

Environmental
Power Adapter: 100-240VAC@DC12V1A
Power Consumption:15W
Operating Temperature:0 ºC  ~ 45 ºC
Storage Temperature: -20 ºC ~80 ºC
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H): 225*150*38mm
Unit Weight: 0.8kg
Compliance: CE, FCC



 

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