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E1 voip Gateway 20 channel SS7 PRI to SIP
BD-20E1-SIP is a new-generation intelligent E1 VoIP gateway which can support 20channels SS7 or PRI to SIP transmission. carrier-grade VoIP and FoIP.

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BD-20E1-SIP is a new-generation intelligent E1 VoIP gateway which can support 20channels SS7 to SIP or PRI to SIP transmission. The 20E1 voip gateway is ideal application for enterprises, telecom operators and various industries. Focusing on a concept of maintainable, manageable  and operable, high integration and large capacity. It provides carrier-grade VoIP and, as well as value-added functions such as  modemand  voice recognition. Thus it constructs a flexible, high-efficient,  future-oriented communication network for users.

BD-20E1-SIP 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 E1 trunk gateway is ideally fit for various access networks of SMEs, call centers, telecom operators and  large-scale enterprises.


    Carrier grade hardware design, 1+1 power supply

    High-integrated structure, up to 20 E1ports 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,Ciscoand ZTE etc.

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: 10/100/1000 BaseT Adaptive Ethernet  GE0: 
     10/100/1000 BaseT Adaptive Ethernet 
Port          * RS232, 115200bps

 Software Features

Local/Transparent Ring Back Tone  Overlapping Dialing

Dialing Ruleswith up to 2000

PSTN group by E1 port or E1Timeslot  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


ISDN PRI  23B+D(T1),30B+D(E1),NT or TE  ITU-T Q.921, ITU-T Q.931, Q.Sig

Signal 7/SS7


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,

Clock : Local/Remote Clock Source             

 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


Web GUI Configuration  Data Backup/Restore  PSTN Call Statistics
  SIP Trunk Call

Firmware Upgrade via TFTP/FTP/Web  Network Capture



Debug, Info, Error, Warning , Notice  Call History Records via Syslog

NTP Synchronization

Centralized Management System

 VoIP Protocol

SIP v2.0 (UDP/TCP),RFC3261  SDP,RTP(RFC2833), RFC3262,


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


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:

Compliance: CE, FCC

 Call Features

Flexible Route Methods


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

Ordering information

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