Voicemaster Voip advanced routing/billing platform for every one!
Preview the SIP routing procedures in action on this demo video:
https://vimeo.com/116332305
I've been working on this system for a while, and now would love to share my expierence and resources of this wonderfull and powerfull Voip platform . The system consist of two servers :
- VoiceMaster VM2000: Billing/Routing ( Kamailio with Sybase database )
- Sysmaster SM7000: Media proccessing ( asterisk with PRI cards )
This system Key Features :
Standard and Advanced VoIP Billing Functionality Advanced Management of Class 4 Service Multiple Authentication Methods Selection of Call Legs for Billing Purposes Real-time Monitoring and Alerts Comprehensive Reporting Virtual Server Partitioning Carrier Grade Reliability Modular Architecture Unlimited Rate Tables High Call Capacity
Here is a few training video in the past, it gives the general overvew of the main functions. Includes a demo for setup PBX Extension , VoiceMail, VirtiualOffice, Conference using Customer Portal, Redundancy, Cloud PBX setup.
https://vimeo.com/user36231282/videos
in 2024 the project still alive and growing. We have running a number of VoiceMaster servers, providing support and security services for them.
- Automatic firewall generation based on information from the database: customer and Voip provider Gateways IP
list dynamically generated and feede for the iptables firewall whitelisting )
- Cluster support for Redundancy - Live replication between Master -> Slave , and automatic takeover.
- DB backups, restore
Its not good idea to run SIP proxy on default port 5060, so we always try to use random ports in setup. If you still want to use a default SIP port: 5060, then it must be closed initially, so only whitelisted IPs has access to it.
Improoved rating system : Provider rate definition:
Customer Rate Definition:
We are opened for new integrations and setups. you can comment or write us via contact-Us form at :