DTU-301 (Digital Trunk Unit) is an open-source asterisk-based VoIP Gateway Module solution for operators and call centers. It is a converged media gateway product, which could be used with OpenVox UCP Series products. This kind of gateway connects traditional telephone system to IP networks and integrates VoIP PBX with the PRI/SS7/R2 seamlessly. With friendly GUI, customers may easily setup their customized gateway. Also secondary development can be completed through API. The DTU-301 supports 1 software-selectable T1/E1 interface and supports up to 30 concurrent calls.
Main Feature
•Providing 1 optional T1/E1/PRI interfaces
•Support up to 30 concurrent calls
•Support most VoIP system platforms, such as Asterisk, Elastix, 3CX, FreeSWITCH ,Broadsoft, etc
Application
•Traditional PBX upgrade to VoIP service
•Traditional PBX connect to remote sites for private VoIP contact
•TLS/SRTP/HTTPS security technology for call and account security
•Configurable E1/T1 port, Support with PRI, SS7, MFC/R2
•IPPBX connect to legacy TDM services
•Phased transition from traditional PBX to IPPBX
•Virtualized system connect to TDM service
•Support multiple voice codes: G.722, G.729, G.723, GSM, G.711
•With advanced echo cancellation technology to ensure high-definition sound quality
•Support T.30, T.38 fax
•Support multi-language voice prompt
•SNMP v1/v2/v3
Panel Appearance
The DTU-301 panel appearance is shown in Figure 1.
Figure 1 DTU-301 Panel Appearance
Parent topic: Module Board