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GSM GATEWAY

4G LTE enabled flexible GSM solution for the Open Source IP PBX & Call Center Dialers

*astTECS 4 Port GSM Gateway is an ideal 4G LTE solution for small businesses or for new Call Centers with small number of agents. This can be connected to one or more lines of IP PBX to a VOIP phone system or provider. This 4 Port GSM VOIP Gateway has 4 channels SIM Card Slots that can handle upto 4 concurrent calls and has ability of implementing calls between 4G/3G/2G network.

*astTECS® GSM VoIP Gateways are purpose built to perform essential GSM VoIP Functions. Optimized for maximum performance and feature integration, *astTECS® Gateways are designed on top of the robust Open Source VoIP Platforms. It is full interoperability with all PBXs, soft-switches and SIP-based environment. 

As VoIP requirements continue to evolve, the processing and I/O requirements for various network devices will also evolve. To meet the demands of ever changing scalability requirements the *astTECS® gateways leverage the reliability of Asterisk which is custom built for Telephony Applications.

Product Highlight:
  • The *astTECS GSM Gateway series includes 3 models with 4, 8, 32 Ports respectively.
  • Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
  • Manageability, a simple configuration, superb voice and video quality and feature rich functionality
  • Based on open industry standards
Voice Capability
  • G.711A/U law, G.723.1, G.729A/B
  • Silence Suppression & Detection
  • Comfort Noise Generation(CNG)
  • Voice Activity Detection(VAD)
  • Echo Cancellation(G.168)
  • Adaptive (Dynamic) Jitter Buffer
  • Call Progress Tone Generation
  • Programmable Gain Control
Physical Interfaces
  • SIM Card Slot: 4/8/32 Channels
  • Ethernet Interface: 2 LAN 10/100M
  • Base-Tx RJ45
  • Console: 1* RS232 115200bps
  • Antennas Connectors: SMA
  • LED Indicators: PWR, RUN, Signal
  • Strength, On/Off- hook
  • Reset Button
  • SIM Card Installation Local
SIP Characters
  • SIP V2.0 RFC3261
  • SDP RFC2327
  • Session Timer RFC4028
  • RTP/RTCP RFC3551
  • SIP Registration
  • SIP Trunk ( Peer Mode)
  • SIP Trunk Group
  • Ring back (Immediately, Alerting)
  • Configurable SIP Release Code
  • Outbound Proxy
  • DTMF Mode: Signal/RFC2833
  • NAT Traversal Dynamic NAT, Static NAT, STUN
Mobile Feature
  • Frequency Range: GSM: 850/900/1800/1900MHz
  • SMS/SMSC/USSD
  • SMS Coding/Decoding: ASCII/UCS2
  • Open API Protocol
  • PIN Code Management
  • Call Duration Limitation for SIM
  • Card/Single Call
  • Polarity Reversal
  • Answer Delay (CDMA)
  • Carrier Selection (GSM)
  • Caller ID Restriction for Outgoing Call
  • BCCH Management
  • Call Waiting
  • Call Forwarding