Description
Dinstar Multi-SIM GSM VoIP Gateway Dubai
The Dinstar Multi-SIM VoIP Gateway Dubai is a robust and high-performance communication solution designed to integrate advanced Multi-SIM technology, featuring four SIM slots for each GSM/3G/4G channel to deliver smooth and uninterrupted connectivity between mobile networks and VoIP systems. By merging the functions of a powerful GSM/3G/4G gateway with a built-in SIMBank, this versatile device offers a flexible, cost-effective, and easy-to-deploy platform that greatly simplifies communication management for organizations of all sizes. It is especially beneficial for enterprises, telecom operators, and bulk SMS service providers that require a reliable, scalable, and efficient system to optimize their communication infrastructure while reducing ongoing operational expenses and maintaining consistent performance across their networks.
Features
Multi-SIM VoIP GSM/3G/4G gateway
- 4 SIM slots per 1 GSM/3G/4G channel
- 32 chs / 128 SIM slots, 16 chs / 64 SIM slots, 8 chs / 32 SIM slots
- Hot swappable SIM cards
- All SIM slots in Front panel, easy to manage SIMs
- Flexible SIMs allocation
- SMS API for bulk SMS application
Applications
- Mobile connectivity for SME IP phone system
- Mobile trunking for Multi-site offices
- GSM/3G/4G as voice backup trunks
- Call termination for service providers
- Land-line replacement for rural area
- Bulk SMS service
- Call Center / Contact Center Solution
Easy Management
- Intuitive Web Interface
- System Log
- Configuration Backup & Restore
- Advanced Debug tools
Technical Highlights
- 4 SIM slots per 1 GSM/3G/4G channel
- Auto CLIP
- 32 chs / 128 SIM slots, 16 chs / 64 SIM slots, 8 chs / 32 SIM slots
- Signaling & RTP Encryption
- Built-in antennas combiner (Optional)
- SMPP for SMS
- GSM: 850/900/1800/1900Mhz
- HTTP API for SMS
- WCDMA: 900/2100Mhz or 850/1900Mhz
- Polarity Reversal
- LTE: Multiple frequency choices for different countries
- PIN Management
- SIP v2.0, RFC3261
- SMS/USSD
- SIM Rotating by SIM run time, SIM balance
- SMS to Email, Email to SMS
- Codecs: G.711A/U , G.723.1, G.729AB
- Call Waiting/Call Back
- Echo Cancellation
- Call Forward
- DTMF: RFC2833, SIP Info
- GSM Audio Coding: HR, FR,EFR, AMR_FR,AMR_HR
- Programmable Gain Control
- HTTPS/HTTP Web Configuration
- Mobile to VoIP, VoIP to Mobile
- Configure Backup/Restore
- SIP Trunk and Trunk Group
- Firmware Upgrade by HTTP/TFTP
- Port and Port Group
- CDR(10000 Lines Storage Locally)
- Caller/Called Number Manipulation
- Syslog/Filelog
- SIP Codes Mapping
- Traffic statistics: TCP,UDP,RTP
- White/Black List
- VoIP Call Statistics
- PSTN/VoIP Hotline
- PSTN Call statistics: ASR,ACD,PDD
- Abnormal Call Monitor
- IVR Customization
- Call Minutes Limitation
- Auto Provisioning
- Balance Check
- SIP/RTP/PCM Capture
- Random Call Interval

