E1 TO SIP CONVERTER 1 Port DGW-L1 Deskripsi :

OpenVox Converter / konverter E1 ke SIP adalah peralatan E1 yang sudah terbukti berjalan mulus saat integrasi ke semua PBX E1 : Avaya, Cisco, Lucent Alcatel, Nortel, Philips, NEC.

Dijamin connect 100% tanpa masalah.

OpenVox DGW-L1 T1 / E1 Gateway adalah solusi Gateway VoIP berbasis asterisk open source untuk operator dan pusat panggilan. Ini adalah produk gateway media terkonvergensi. Jenis gateway ini menghubungkan sistem telepon tradisional ke jaringan IP dan mengintegrasikan VoIP PBX dengan ISDN secara mulus. Dengan GUI yang ramah, pengguna dapat dengan mudah mengatur Gateway khusus mereka. Pengembangan sekunder juga dapat diselesaikan melalui AMI (Asterisk Management Interface).

DGW-L1 T1 / E1 Gateway mendukung 1 interface T1 / E1 / PRI yang dapat dipilih perangkat lunak dan mendukung hingga 30 panggilan bersamaan.

E1 Gateway OpenVox kualitas Premium, dimana OpenVox sudah terkenal karena kualitas produk kelas premium.  Untuk E1 Gateway disarankan memakai produk kelas premium, karena alat ini berefek secara luas terhadap System Telekomunikasi Customer.

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DEMO ONLINE : ( Anda dapat melihat langsung secara LIVE isi program dan setting )

http://demo.openvox.cn:65327

admin / admin

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 T1/E1 RJ-48

2 10/100/1000Mbps Ethernet ports

2 USB 2.0 ports

Maximum Power Consumption: 12W

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 port T1/E1, energy efficiency concurrent processing, up to 30

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 for custom dialplans

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

Good compatibility, support Asterisk, Elastix, Freeswitch and Small and medium IPPBX platform

SIP Features :

Support add, modify & delete SIP Accounts

SIP registration with Domain

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