Easy to use - graphical user interface

• Nice and easy to use graphical user interface to view live calls data and call status , also ASR/ACD live monitoring is available
• Everything in our solution is user friendly and can be managed easily through the web interface
• No need to have difficult skills to manage the system like writing scripts for routing calls .. ETC
• Flexible design you can run the system from any PC web browser or Mobile browser

Data and Reporting

• Detailed statistics by hour , day , or any period of time
• Real-time system status dashboard: ASR, ACD, current calls
• Real time Credit/Balance reconciliation
• Active Calls Monitoring
• Real time balance update
• Users reports with full CDR
• Service provider costs and net profit calculations reports
• View and manage client invoices report

Setup and Configurations

• Few and easy steps to manage your clients , routings , and gateways
• Unlimited clients (originators) wholesale and retail , gateways/trunks , tariffs , and routing plans
• Flexible routing plans , you can set multiple destinations for same prefix , the system will re-route the call to next destination that associated to same prefix in case if first destination unable to terminate the call
• Save your time by enable our invoicing module , the system will issue and send automatically an invoice for each client weekly
• Filter incoming VoIP traffic based on dialed number , or caller id , also it support white and black lists , static and dynamic lists , for more details please check our guide
• Support Http lookup api to validate dialed numbers on external third party before call hit your destination/gateway in real time , for more details please check our guide

Server and Technical Specs

• Compatible with Android phones GSM gateway like Pure-VoIP.com , and other SIP devices and trunks
• Automatic codec conversion/transcoding
• Supported codecs are GSM, iLBC, G711u-Law, G711a-Law, and G729
• Dedicated and secure resources , and dedicated IP Address
• Manage system time zone based on needed timing to avoid any conflict with your clients
• Media Timeout : this feature will track the voice stream RTP in the call sessions and it will drop the call if the RTP is not received for a period of time , this feature will ensure the accuracy for your calls duration and invoices , this feature is handy because sometimes the SIP endpoints for certain network errors are not sending BYE message to hang-up the connected calls

Multi Security Levels

• Maximum security against unauthorized login or hacking attempts.
• Auto block of hacking attempts
• IP auth to manage who can send calls through the system
• Intelligent algorithm to detect suspicious network behaviors and stop it
• Built in high performance firewall

Customizing

If your needed feature is not available in our solution , we can add custom features on the solution to match your business needs