20211230024327191

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.

INFORMATION

Overview

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 FoIP.services, 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.

Key Features

•    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 
Serial Port          * RS232, 115200bps

 Software Features

Local/Transparent Ring Back
Tone  Overlapping Dialing

Dialing Rules,with 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

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

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

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


Maintenance

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

Firmware Upgrade via TFTP/FTP/Web  Network Capture

SNMP v2

Syslog:

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,
  3263,3264,3265,3515,2976,3311

SIP TLS/SRTP

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

Environmental

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

 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


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