OpenVox VS-GW2120-12G

OpenVox VS-GW2120-12G gateway - high-density VoIP-GSM gateway for 12 channels with the possibility of expansion to 44 channels, is industrial equipment for small, medium and large businesses.

The OpenVox VS-GW2120-12G gateway is a high-density 12-channel VoIP-GSM gateway with the ability to expand to 44 channels, is industrial equipment for small, medium and large businesses. Thanks to the user-friendly graphical web interface and unique modular design, users can not only easily configure their gateway, but also scale the device, expanding it as the company grows. Also, additional functionality can be achieved using AMI (Asterisk Management Interface). The new OpenVox VS-GW2120-12G VoIP-GSM gateway is made in a 2U chassis format and has a modular structure, providing placement of up to 11 built-in 12-channel GSM modules. The gateway is equipped with 3 Ethernet ports and supports G.711A, G.711U, G.729, G.722, G.723, G.726 and GSM codecs. Due to its unique design, the OpenVox VS-GW2120-12G gateway supports hot swapping of both SIM cards and GSM modules. Users can simply add or remove modules as needed, expanding the hardware capabilities of the gateway. Each GSM module has its own independent operating system based on Asterisk software. The gateway supports SMS sending, receiving, bulk sending and SMS to email conversion, and is 100% compatible with Asterisk, Elastix, Trixbox, 3CX FreeSWITCH Sip Server and VOS VoIP platforms.
Main features of OpenVox VS-GW2120-12G: 12 GSM channelsExpandable to 44 GSM channels1 SIM card per 1 GSM channel3 LAN interface 10 / 100MHot swapping of SIM cards and GSM modules OpenVox VS-GW2120-12G functions: Simple and convenient configuration via web interfaceSupport for multiple interface languagesCodecs: G.711A, G.711u, G.729, G.723.1, G.722, G.726, GSMGSM: 850/900/1800/1900 MHzSIP (RFC3261) complianceSupport for various SIP protocols, IAX2 (coming soon)DTMF signaling support: RFC2833, Inband, SIP InfoInterface API interface (AMI)Echo cancellation: ITU-T G.168 LECSupport static IP and DHCPSupport OpenVPN and PPTP (coming soon)Limited call durationCLIR - Caller ID restrictionCall waitingCall forwarding (unconditional, no answer, busy, not available)SMS main radio station, SMS automatically redirectHTTP interface for SMS managementSMS to E-mailUSSD transceiverSMS remote gateway controlAutomatic IME changeIPIN identificationMobile Number Portability (MNP)Flexible routing settings and unlimited routing rulesVarious applications such as IVR, DISASupport configuration file backupSupport user scripts, DialPlansLeast Cost Routing (LCR), depending on time, port, caller IDIndependent system for each moduleAffordable price and excellent performanceHot swapping of SIM cards and modulesCompatible with Asterisk, Elastix, 3CX, FreeSWITCH SIP servers.