20E1 to SIP Voip Trunk Gateway

20E1 to SIP Voip Trunk Gateway

Short Description:

Overview BD-MTG-20E1 is a new-generation intelligent VoIP gateway, which is designed for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable and operable, the gateway features high integration and large capacity. It provides carrier-grade VoIP and FoIP. services, as well as value – added functions such as modem and voice recognition. Thus it constructs a flexible, high-efficient, future-oriented communication network for users. The VOIP gateway supports a range of signaling protocols, realizing the interconnection between SIP and traditional signals like SS7 and PRI. It supports multiple codec methods, offers signal encryption technology and smart voice recognition…
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  • Description

    Specification

    Application

    Order information

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    Description

    Overview

    BD-MTG-20E1 is a new-generation intelligent VoIP gateway, which is designed for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable and operable, the gateway features high integration and large capacity. It provides carrier-grade VoIP and FoIP. services, as well as value – added functions such as modem and voice recognition. Thus it constructs a flexible, high-efficient, future-oriented communication network for users.

    The VOIP gateway supports a range of signaling protocols, realizing the interconnection between SIP and traditional signals like SS7 and PRI. It supports multiple codec methods, offers signal encryption technology and smart voice recognition technology, and improves the utilizing efficiency of trucking resources while ensuring voice quality. The trunk gateway is ideally fit for various access networks of SMEs, call centers, telecom operators and large-scale enterprises.

    Key Features

    • Carrier grade hardware design, 1+1 power supply
    • High-integrated structure, up to 20 E1 ports in 1U size
    • Support flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments
    • Support multiple coding standards: G.711A/U, G.723.1, G.729A/B and iLBC
    • High compatibility, interoperable with PBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei,Cisco and ZTE etc.
    • Modular design, the capacity can easily adjusted by adding/reduce E1 cards (4E1 per card), reducing project cost.

     

    Physical Interfaces

    PSTN

    Software Features

    E1/T1 Ports
    ISDN PRI
    Local/Transparent Ring Back Tone
    4/8/12/16/20 E1/T1 23B+D(T1),30B+D(E1),NT or TE

    ITU-T Q.921, ITU-T Q.931, Q.Sig

    Overlapping Dialing
    DTU Module:
    Signal 7/SS7 Dialing Rules, with up to 2000
    4 E1/T1 ITU-T, ANSI,ITU-CHINA

    MTP1/MTP2/MTP3, TUP/ISUP

    PSTN group by E1 port or E1 Timeslot
    Interface Type
    E1 Frame Type

    DF,CRC-4,CRC_ITU

    IP Trunk Group Configuration
    RJ48(Impedance 120Ω) T1 Frame Type Voice Codecs Group
    Ethernet Interface
    4-Frame Multi-frame (F4,FT), 2-Frame Multi-frame (F12, D3/4),  Extended Super-frame (F24, ESF) ,  Remote Switch Mode (F72, SLC96) Caller and Called Number White Lists
    GE1: 10/100/1000 BaseT Adaptive Ethernet Line Codes Caller and Called Number Black Lists
    GE0: 10/100/1000 BaseT Adaptive Ethernet E1:NRZ,CMI,AMI,HDB3 Access Rule Lists
    Serial Port
    T1:NRZ,CMI,AMI,B8ZS IP Trunk Priority
    1* RS232, 115200bps Clock: Local/Remote Clock Source

     

    Voice Capabilities Maintenance VoIP Protocol
    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

    Web GUI Configuration

    Data Backup/Restore

    PSTN Call Statistics

    SIP Trunk Call Statistics

    Firmware Upgrade via TFTP/FTP/Web

    Network Capture

    SNMP v2

    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

    Environmental Syslog

    SIP/IMS Registration

    With up to 2000 SIP Accounts

    NAT: Dynamic NAT, Rport

    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

    Debug, Info, Error, Warning , Notice

    Call History Records via Syslog

    NTP Synchronization

    Centralized Management System

    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

    Specification
    Application
    Order information
    Q&A

    Question1: Can I add static routes in E1 to SIP gateway?
    Answer: Yes, it can be added.

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