E1 TO SIP CONVERTER OPENVOX DGW-1002 DESKRIPSI :

OpenVox T1 / E1 Gateway adalah solusi VoIP Gateway berbasis open source untuk operator dan pusat panggilan. OpenVox E1 Gateway adalah produk gateway media konvergen. Gateway semacam ini menghubungkan sistem telepon tradisional ke jaringan IP dan mengintegrasikan VoIP PBX ke ISDN dengan mulus. Dengan GUI yang mudah digunakan, pelanggan dapat dengan mudah mengatur Gateway sesuai kebutuhan. Dan pengembangan sekunder dapat diselesaikan melalui AMI (Asterisk Management Interface).

Gateway VoIP digital DGW-100X (R) mendukung 1/2/4 antarmuka T1 / E1 / PRI yang dapat dipilih oleh perangkat lunak dan mendukung hingga 30/60/120 panggilan serentak. "R" berarti perangkat mendukung catu daya redundan.

Target Applications

Connect legacy PBX systems to low-cost VoIP services

Connect legacy PBX systems to remote sites over private VoIP links

Connect IP PBX systems to legacy TDM services

Phased transition from legacy PBX to IP PBX

Connect virtualized systems to legacy TDM services

Transcoding by connecting systems using varying codecs

Lync connectivity to SIP or legacy TDM providers and SIP or Legacy PBX

Technical Specifications

1/2/4 T1/E1  RJ-48

2 10/100/1000M Ethernet ports

2 USB 2.0 ports

DGW-100XR with redundant power supply

DGW-100X with single power supply

Maximum Power Consumption: 20W

Power supply specification: 100-240V/AC

Operation humidity: 5%~95% non-condensing

Operating temperature: 0℃~70℃

Storage temperature: -40℃~85℃

Features

System Features

Available in 1/2/4 port T1/E1, energy efficiency concurrent processing, up to 120

Signalling: PRI/R2/SS7

Support up to 24 countries’ standard R2 signalling

Support new R2 variant

Simple and convenient configuration via Web GUI

Codecs support: G.711A, G.711U, G.729A, G.723.1, G.722, GSM

Support protocols：SIP、IAX、TCP、UDP、RTP、SSH、HTTP、HTTPS

Support NTP time synchronization and client time synchronization

Support SSH access for background management, Asterisk CLI command operation

Open API interface (AMI)

Support ports group management

Support custom dialplans

Echo Cancellation (Octasic® DSP)

Firmware update by HTTP

Support call statistics

Support auto provision

Support channel status show dynamically

Support backup/upload configuration file

Multiple detailed log output

Support Chinese language

Automatically reboot

Compatibility with all kinds of SIP servers, such as Asterisk, Elastix, Freeswitch and other IPPBX platforms

Available for OEM

3-month "No Question Asked" Return Policy, and Two-year Warranty

SIP Features

Support add, modify & delete SIP Accounts

SIP registration with Domain

Support multiple SIP registrations：Anonymous, Endpoint registers with this gateway and this gateway registers with the endpoint

SIP accounts can be registered to multiple servers

Combine different SIP Trunks into group

SIP(RFC3261) compliance

DTMF: RFC2833, SIP INFO, INBAND

Support T.38 /Pass-through Fax

Routing

Flexible routing settings

Support 512 routing

Support caller/callee manipulation and filtering

Trunk group support, Trunk priority management

Support add, modify & delete routing

E1/T1 port grouping

Support Failover

Network Features

Network type: Static IP and DHCP

IPv4, UDP/TCP, DHCP, TFTP, SCP

HTTP/HTTPS/SSH

Support DDNS

Support ping & traceroute command on the web

Support network capture on the web

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