Development marches on! 2.0.0 released.

2008-01-20 20:07:53

All this time development proceeded non-stop. As a result, there have been so many changes made, that we have decided to increment release major-number.

Here is the list of the essential changes:

• Routing core was refactored. Inner inter-module interfaces were changed to serve the calls as fast as possible.
• IVR functionality has been integrated into the system. The system copies cisco debit card application fuctionality, including AAA, balance auding, time left auding, etc. Capabilities are very similar to those provided by cisco, but there is one small difference - you don't have to buy expensive equipment from cisco to arrange the calling cards scheme. Why do you need to pay more? Additionally interface is very simple and extremely user-friendly due to integration into product mainline.

  Points of entrance for IVR incoming calls are configured in IVR Access Points. Here you can configure signaling options of the PSTN gateway, from whom the calls are awaited. Also you can setup authentication parameters here (secret code and ANI(calling number) authentiction is supported) and customize phrases that are said.

  Authentication, routing and billing are performed basing on the end users settings.

• Changes in naming convention and file paths.
• Logging politics change: from now on asterisk VoIP protocol logs are kept in /var/log/asterisk, where they are periodically rotated; softswitch logs, including SQL queries, are maintained in /usr/local/softswitch. This gives us the opportunity to restore data in horrible case of database engine failure.

Thank you for your support!