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!

Changelist:

• Payments & Invoices logic was changed;
• System moved to Asterisk 1.4.14;
• Some minor bugs were fixed.

At last the User's Manual was written. It is still not completed, but it's better than nothing. We recommend starting usage of the system step-by-step according to the instructions from the Getting Started section .

End users are able to accept incoming calls now!

End users have no permanent IP address according to system's logic, so they must periodically register on the server to let the VoIP-core know, where the call must be routed. This can be configured on IP phone or dialer.

From the Switchboard side you must add one or few source number(s) to the end user. Routing Module transforms the called number to the "routed destination number" , then it determines the end user to which the call must be routed. You can also specify in Routing Strategies for inbound dial-peer 'did' as the most preferred method of route comparison. In case you don't, the call destinated to the end user, could be routed to provider, which provides matching country code.

Incoming calls can be billed basing on the separate pricelist.

This functionality makes possible to sell DID(Direct Inward Dial) numbers to the end users and billing such calls.

This release contains several minor fixes in web-interface. Also following features have been added:

• The system now supports SIP end users. This makes possible to use the system in PC2Phone business application.

  End users should authenticate themself by password (which is stored as plain-text or MD5-encrypted password in system) before they are able to call. End users are gathered into groups. Each group has its own routing settings and pricelist.

  End user's call origination can be prohibited by 2-stage balance checking strategy. First, end user's balance is checked. Second, company's (to which the end user group belongs) balance is checked. End user balances and CDR are maintained for each user individually and can be viewed in real-time mode.

• Some minor changes in web-interface to improve usability. Quick links to Statistics and Dialplan from dial-peers have been added.
• Some minor changes in web-interface to improve its look.