DINSTAR UC2000-VE GSM/WCDMA/LTE VoIP Gateway
DINSTAR UC2000-VE GSM/WCDMA/LTE VoIP Gateway

DINSTAR UC2000-VE GSM/WCDMA/LTE VoIP Gateway

The DINSTAR UC2000-VE is a compact and reliable hardware solution that serves as a 4 or 8 channel GSM/3G/4G VoIP Gateway. Designed to seamlessly bridge mobile and VoIP networks, it enables the transmission of voice and SMS communications. With integrated GSM/WCDMA/LTE connectivity and compatibility with the SIP protocol, which is widely supported by major VoIP platforms, the UC2000-VE is an ideal choice for enterprises, multi-site organizations, call terminators, and areas with limited landline infrastructure, such as rural regions. By leveraging this gateway, organizations can significantly reduce telephony costs and facilitate effortless and efficient communication. The UC2000-VE can operate as a standalone device or support remote SIM management when used in conjunction with the Dinstar SIMBank and SIMCloud.

Key Features
    • GSM/WCDMA/LTE Support
      Comprehensive support for GSM, WCDMA, and LTE networks.
    • Voice over LTE (VoLTE)
      High-quality voice calls over LTE networks.
    • Hot Swappable SIM Cards
      Easily replace SIM cards without interrupting operation.
    • Compatible with Mainstream VoIP Platforms
      Seamless integration with leading VoIP platforms.
    • Mobility Extension, Never Miss a Call
      Stay connected on-the-go, never miss important calls.
    • SMS Sending & Receiving, SMS API
      Effortless SMS messaging with SMS API integration.
    • Credit Limit Management
      Efficient control and monitoring of telephony expenses.
    • Auto CLIP
      Automatic Calling Line Identification Presentation (CLIP) for incoming calls.
  • VoLTE
    VoLTE
  • Voice
    Voice
  • SMS
    SMS
  • API
    API
  • SIP
    SIP
  • SIMCloud
    SIMCloud
  • HD Audio

Application

  • Mobile Connectivity Solution for SME IP Phone Systems:  Connect your SME IP phone system to mobile networks for enhanced mobility and communication.
  • Mobile Trunking Solution for Multi-site Offices:  Enable mobile trunking across multiple office locations to streamline communication and improve efficiency.
  • GSM/3G Voice Backup Trunks:  Utilize GSM/3G networks as backup trunks for voice communication, ensuring uninterrupted connectivity.
  • Call Termination Solution for Service Providers:  Efficiently terminate calls for service providers, enabling seamless communication for their customers.
  • Land-line Replacement Solution for Rural Areas:  Replace traditional land-line infrastructure in rural areas with our solution, extending reliable communication services.
  • Bulk SMS Service:  Leverage our bulk SMS service for sending high-volume SMS messages quickly and easily.

Streamlined Administration: Ease of Management

Empowering Connections, Enhancing Business
  • Intuitive Web Interface:  Effortlessly navigate and manage the system through a user-friendly web interface.
  • Remote SIMs Management with Dinstar SIMBank & SIMCloud:  Remotely manage SIM cards using the Dinstar SIMBank and SIMCloud solutions for seamless control and flexibility.
  • Configuration Backup & Restore:  Safeguard your configurations by easily creating backups and restoring them as needed.
  • Advanced Debug Tools:  Access powerful debugging tools for efficient troubleshooting and diagnostics.

Features

Enhance Communication with Advanced Features
    • 4/8 SIM Slots and 4/8 Antennas:  Flexible configuration with 4/8 SIM slots and 4/8 antennas for enhanced connectivity.
    • Built-in Antennas Combiner (Optional):  Optional feature that combines built-in antennas for improved signal reception and performance.
    • GSM: 850/900/1800/1900MHz:  Support for GSM frequencies including 850/900/1800/1900MHz for global compatibility.
    • WCDMA: 900/2100MHz or 850/1900MHz:  WCDMA support with options for frequencies 900/2100MHz or 850/1900MHz, ensuring wide coverage.
    • LTE: Multiple Frequency Choices for Different Countries:  LTE compatibility with multiple frequency options to cater to various countries and regions.
    • SIP v2.0, RFC3261:  SIP protocol support with version 2.0 compliance according to RFC3261 standard.
    • Codecs: G.711A/U, G.723.1, G.729AB:  Wide range of supported codecs including G.711A/U, G.723.1, and G.729AB for efficient audio compression and decompression.
    • Echo Cancellation:  Built-in echo cancellation feature to eliminate echo and ensure clear audio during calls.
    • DTMF: RFC2833, SIP Info:  Support for DTMF signaling using both RFC2833 and SIP Info methods.
    • Programmable Gain Control:  Adjustable gain control for fine-tuning audio levels and optimizing sound quality.
    • Mobile to VoIP, VoIP to Mobile:  Seamlessly connect mobile networks to VoIP networks and enable communication between the two.
    • SIP Trunk and Trunk Group:  Set up SIP trunks and trunk groups for efficient routing and management of incoming and outgoing calls.
    • Port and Port Group:  Configure individual ports and group multiple ports together for convenient call management.
    • Caller/Called Number Manipulation:  Manipulate caller and called numbers for advanced call routing and customization.
    • SIP Codes Mapping:  Map specific SIP codes to desired actions or behaviors for call handling and troubleshooting purposes.
    • White/Black List:  Create white and black lists to control incoming and outgoing calls based on specific numbers or criteria.
    • PSTN/VoIP Hotline:  Set up hotlines for seamless communication between PSTN (Public Switched Telephone Network) and VoIP networks.
    • Abnormal Call Monitor:  Monitor and detect abnormal call patterns or behaviors for enhanced security and fraud prevention.
    • Call Minutes Limitation:  Set limitations on call duration or total call minutes for better cost control and resource management.
    • Balance Check:  Check the remaining balance or credit for prepaid services to ensure efficient usage and avoid service interruptions.
    • Random Call Interval:  Configure random call intervals to simulate human call patterns and enhance call authenticity for certain applications.
    • Auto CLIP:  Automatic Calling Line Identification Presentation (CLIP) for incoming calls to display caller information.
    • Signaling & RTP Encryption:  Secure signaling and RTP (Real-time Transport Protocol) encryption for enhanced privacy and data protection during communication.
    • SMPP for SMS:  Support for SMPP (Short Message Peer-to-Peer) protocol for efficient and reliable SMS messaging.
    • HTTP API for SMS:  Utilize HTTP API for seamless integration and programmable control of SMS messaging functionalities.
    • Polarity Reversal:  Polarity reversal support for compatibility with specific telephone line configurations.
    • PIN Management:  Manage PIN (Personal Identification Number) codes for enhanced security and access control.
    • SMS/USSD:  Send and receive SMS messages and USSD (Unstructured Supplementary Service Data) codes for interactive communication.
    • SMS to Email, Email to SMS:  Convert SMS messages to emails or emails to SMS messages for seamless integration and communication across platforms.
    • Call Waiting/Call Back:  Support for call waiting and call back features to enhance call management and user experience.
    • Call Forward:  Configure call forwarding to redirect incoming calls to other destinations or numbers.
    • GSM Audio Coding: HR, FR, EFR, AMR_FR, AMR_HR:  Support for various GSM audio coding standards, including HR, FR, EFR, AMR_FR, and AMR_HR for optimized audio quality.
    • HTTPS/HTTP Web Configuration:  Secure HTTPS and standard HTTP web-based configuration options for easy device setup and management.
    • Configure Backup/Restore:  Easily create backups and restore device configurations for hassle-free setup and maintenance.
    • Firmware Upgrade by HTTP/TFTP:  Upgrade firmware using HTTP or TFTP (Trivial File Transfer Protocol) for seamless device updates and enhancements.
    • CDR (10000 Lines Storage Locally):  Store call detail records (CDR) locally with a capacity of up to 10,000 lines for easy tracking and analysis of call data.
    • Syslog/Filelog:  Capture and store system logs and file logs for comprehensive system monitoring and troubleshooting.
    • Traffic Statistics: TCP, UDP, RTP:  Collect and analyze traffic statistics for TCP (Transmission Control Protocol), UDP (User Datagram Protocol), and RTP protocols.
    • VoIP Call Statistics:  Track and analyze statistics related to VoIP calls, including call volume, duration, and quality metrics.
    • PSTN Call Statistics: ASR, ACD, PDD:  Measure and monitor PSTN call statistics, including ASR (Answer Seizure Ratio), ACD (Average Call Duration), and PDD (Post Dial Delay).
    • IVR Customization:  Customize Interactive Voice Response (IVR) menus and prompts for personalized and efficient call routing and interaction.
    • Auto Provisioning:  Automate device provisioning and configuration processes for streamlined deployment and management.
    • SIP/RTP/PCM Capture:  Capture and analyze SIP, RTP, and PCM (Pulse Code Modulation) data for in-depth troubleshooting and performance optimization.
    • Work with Dinstar SIMCloud/SIMBank (Optional):  Optional compatibility with Dinstar SIMCloud and SIMBank solutions for enhanced SIM management and control capabilities.
Clearing Doubts, Enhancing Experiences

Frequently Asked Questions

Question 1: What is the maximum number of SIM slots and antennas supported by the product?

Answer: The product supports either 4 or 8 SIM slots and 4 or 8 antennas, providing flexible configuration options.

Question 2: Is there an option for built-in antennas combining?

Answer: Yes, the product offers an optional built-in antennas combiner feature for improved signal reception and performance.

Question 3: What are the supported GSM and WCDMA frequencies?

Answer: The product supports GSM frequencies of 850/900/1800/1900MHz and WCDMA frequencies of either 900/2100MHz or 850/1900MHz.

Question 4: Are there multiple LTE frequency options available for different countries?

Answer: Yes, the product offers multiple frequency choices for LTE to accommodate different countries and regions.

Question 5: Which SIP protocols and codecs are supported by the product?

Answer: The product supports SIP v2.0 (RFC3261) protocol and codecs such as G.711A/U, G.723.1, and G.729AB for efficient audio compression.

Question 6: Does the product have built-in echo cancellation?

Answer: Yes, the product is equipped with built-in echo cancellation to ensure clear audio during calls.

Question 7: What DTMF signaling methods are supported by the product?

Answer: The product supports DTMF signaling using RFC2833 and SIP Info methods.

Question 8: Can I customize the IVR menus and prompts?

Answer: Yes, the product allows customization of Interactive Voice Response (IVR) menus and prompts for personalized call routing.

Question 9: Is there a web-based interface for configuration and management?

Answer: Yes, the product provides an intuitive web interface for easy configuration and management.

Question 10: How can I upgrade the firmware of the product?

Answer: The product supports firmware upgrades through HTTP or TFTP protocols for convenient updates.

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    114 Lavender Street, #07-51, CT Hub 2, Singapore 338729
  • Phone Number:
    +65 8757 7860
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  • Email:
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