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OpenVox GW1202 Wireless Gateway

The Wireless gateway is designed with a Lan Switch board that allows for hardware stackability. It allows you to send and receive SMS messages, as well as send and receive group SMS messages and SMS to email. The OpenVox GW1202 Wireless Gateway supports most VoIP operating platforms, including Asterisk, Elastix, 3CX, FreeSWITCH, Broadsoft, and others. It uses the standard SIP protocol and is compatible with Leading IMS/NGN platforms, IPPBX, and SIP servers.

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The OpenVox GW1202 Wireless Gateway, a VS-GW1202 Series upgrade product, is now the leading open source Asterisk-based VoIP Gateway solution for SMBs and SOHOs. Users can easily configure their customized Gateway thanks to a user-friendly interface and a one-of-a-kind modular design. AMI can also be used to complete secondary development (Asterisk Management Interface). The new VoxStack series Gateway comes in three models: the VS-GW1202 V2, the VS-GW1600 V2, and the VS-GW2120 V2 Wireless Gateway.

The OpenVox GW1202 Wireless Gateway with Modular Design can provide up to 2 plug-in GSM/WCDMA/LTE gateway modules that can support from 4 to 8 GSM/WCDMA/LTE channels, developed for interconnecting a wide range of codecs, including G.711A, G.711U, G.729, G.722, G.723, G.726 & GSM, to 2G/3G/4G networks to quickly reduce The OpenVox GW1202 Wireless Gateway’s unique design allows it to support hot-swap for both SIM cards and GSM/WCDMA/LTE wireless gateway modules. Users can simply add or remove modules to expand or exchange hardware. Each GSM/WCDMA/LTE wireless gateway module contains its own asterisk system.


  • Configuration is simple and convenient thanks to the Web GUI.
  • HTTP / HTTPS / SSH are supported.
  • Fundamental security strategy (white/blacklisting, security rules).
  • GSM Voice Codec (optional).
  • G.711A, G.711U, G.729, G.723.1, G.722, G.726, GSM are all supported codecs.
  • Compliance with SIP (RFC3261).
  • Support for various SIP and IAX2 protocols.
  • There are three SIP/IAX2 registration models: none, server, and client.
  • RFC2833, Inband, and SIP INFO all support DTMF signaling.
  • The true Open API Protocol (based on Asterisk).
  • Static IP addresses and DHCP are supported.
  • Call Duration Restrictions.
  • The acronym CLIR is an abbreviation for (Call Line Identification Restriction).
  • Waiting for a call.
  • Call Rerouting (unconditional, no reply, busy, not reachable).

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