OpenVox VS-GW2120-44G

OpenVox VS-GW2120-44G gateway is a high-density VoIP-GSM gateway for 44 channels, which is industrial equipment for small, medium and large businesses.

The OpenVox VS-GW2120-44G gateway is a high-density 44-channel VoIP-GSM gateway, which is industrial equipment for small, medium and large businesses. With a user-friendly graphical web interface and a 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-GW2120-44G VoIP-GSM gateway is made in a 2U chassis format and has a modular structure, providing placement of up to 11 built-in 44-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-GW2120-44G 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, mass mailing 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-GW2120-44G: 44 GSM channels 1 SIM card per 1 GSM channel 3 LAN interfaces 10 / 100M Hot swappable SIM cards and GSM modules OpenVox VS-GW2120-44G functions: Simple and convenient configuration via web interface Support for multiple interface languages Codecs: G.711A, G.711u, G.729, G.723.1, G.722, G.726, GSM GSM: 850/900/1800/1900 MHz SIP (RFC3261) compliance Support for various SIP protocols, IAX2 (coming soon) Support for DTMF signaling: RFC2833, Inband, SIP Info Interface API interface (AMI) Echo cancellation: ITU-T G.168 LEC Support for static IP and DHCP Support for OpenVPN and PPTP (coming soon)Limited call durationCLIR - Caller ID RestrictionCall WaitingCall Forwarding (Unconditional, No Answer, Busy, Not Available)SMS Primary Radio, SMS Automatically ResentHTTP Interface for SMS ManagementSMS to E-mailUSSD TransceiverSMS Remote Gateway ManagementAutomatic IME ChangeIPIN IdentificationMobile Number Portability (MNP)Flexible Routing Settings and Unlimited Routing RulesVarious Applications such as IVR, DISASupport for Configuration File BackupSupport for Custom Scripts, DialPlansLeast Cost Routing (LCR), Time-Based, Port-Based, Caller ID-BasedIndependent System for Each ModuleAffordable Price and Excellent PerformanceHot Swapping of SIM Cards and ModulesCompatible with Asterisk, Elastix, 3CX, FreeSWITCH SIP Servers.