Interconnection has been done using the SIP protocol. Cluster performs load balancing using DNS round-robin coupled with SIP message "302 Moved Temporarely". The nodes that handle media streams work separately from the nodes that handle signalization.

We are glad to announce that our system handles such a complex interconnection scheme and performs proper billing of the outbound calls.

To implement the valid billing we had to patch Asterisk, therefore you won't see this functionality in the original Asterisk.

We are glad to introduce a new, 6.1, version of the Smartswitch system

Main theme of this release is bug fixes. Here is the list of main changes:

• Fixed memory leak when handling CDR. Issue is relevant for the users of 6.0 that handle more than 100 simultaneous calls;
• Several bugs in Asterisk 1.6 have been fixed. This is relevant for users of 6.0, that have CPU consumption about 100%;
• Fixed issue of H323 server that is bind to several IP addresses. Issue is relevant to users with multi-homed setups of all Smartswitch versions;
• Fixed SendMail application. Relevant to users of 6.0;
• Fixed issue of CDR export to file. Relevant to the users of STANDARD edition;
• Fixed issue when Call Handler is launched within Java 6 Update 20. Relevant to users of all Smartswitch versions, starting from 5.0;
• New report has been added - Media Format Report.

  This report enables you to to observe statistical data regarding media streams, that pass through the system, including the percentage of calls with codec translation. This tools allows you to diagnose problems with system performance and media stream quality;

• A new tab has been added to Call Detail Report - RTP statistics.

  This enables you to see parameters of each media stream (packet count, lost packet count, jitter, etc) that passed through the system and quickly react to customer's complains related to voice quality;

• Added feature of quick navigation over the linked CDR in the Call Detail Report;
• Added new option to display free calls separately in Code Report (relevant to users of the Consumption profile feature);
• Added feature of caller ID registration to user's web portal;
• Profit Controllers feature implemented.

  From now on you can precisely control your profit when call is being routed. You can set minimum absolute or relative per-minute profit for your peers. If configured, routing of calls will be performed only to those routes that satisfy these configured profit values;

• Look and feel enhancements.

This release contains significant amount of features and fixes comparatively to previous 5.3 release. Some of the changes are heavy, therefore we decided to increase the major number to 6.

The list of most significant changes:

• The open-source software that is included into the distribution has been updated:
  • FreeBSD to 7.3;
  • Asterisk to 1.6.0.26;
  • MySQL to 5.1.45.
• Added a russian user's guide;
• Implemented access server on the basis of XML-RPC protocol to enable linkage of the system with third-party applications (for example, with web-store with dynamic user registration);
• Web-interface changes:
  • Added the role-based access permission system. You can flexibly assign authorities to each employee of your company;
  • Automatic channels and sessions pages reload has been implemented. You can configure the period of reload;
  • Web-interface changes logging has been implemented. This enables you to see who and when did each and every change in the configuration.
  • An ability to configure the widget classes have been added. This enables you to define custom web-interface control options for administrator and customer web-interface (for example, redirection configuration options);
  • An ability to refill with the PIN code from the subscriber's web portal has been added.
• Media handling changes:
  • Options for control for the Differential Services field in the IP packets for RTP streams has been added. More likely that the voice quality of the voice stream going through the system will significantly improve after this change;
  • Added support for NAT, RTCP and UDPTL (T.38 faxes) in Media Proxy;
  • Media Proxy can be enabled or disabled per each peer independently (on the contrast to the global setting that was in 5.3).
• SIP changes:
  • new options for registration;
  • enhanced support for T.38 faxes(due to migration to 1.6 branch of asterisk).
• H323 changes:
  • added NAT support;
  • added possibility of working with peers via different network interfaces;
  • fixed several critical vulnerabilities in openh323 library, which lead to sporadic call hangup on the load over 100 calls.
• several critical vulnerabilities in asterisk have been fixed. Digium has already committed some of them into their mainline;
• n-way conferences with dynamic participant gathering and correct billing has been implemented;
• Billing changes:
  • a core engine has been refactored to support further enhancements;
  • safe synchronization of transactions have been implemented (to protect situations when someone might want to compromise the system doing actions like transferring funds from one account to another);
  • Proper online balance control in the real-time mode for answered channels has been implemented. This solves a billing issue "to hangup when balance is 0" for the calling cards platform when there are several simultaneous calls from one customer. This feature also might be useful when working with the wholesalers on the prepaid basis;
  • a calendar write-off method (making write-off for all the customers at the same specified time, for example, at 00:00 of the first day of month) has been implemented
  • an option to configure a credit for company of retail customer has been added;
  • new feature: consumption profiles. You can configure the amount of minutes or the money amount for particular destination set and for particular time period. And you can configure billing for this set of rules to differ from the regular one. This enables you to configure rules like "1000 minutes per month for Australian destination codes are free of charge";
  • You can configure billing to not to alter the balance when the call is over (this is mainly used with consumption profiles).
• IP-routing configuration has been added to web-interface;
• An option to configure and control periodic processes in the system has been added (for example, you can change periodic write-off period or system load graphs update period);
• Added MP3 support for file and music on hold playback;
• One more step towards virtual PBX: an option to configure custom sound files for the retail customer that overwrite generic ones;
• Internet-billing implementation has been started: a VPN-server MPD has been added to distribution and linked with the Smartswitch. VPN-server is used to connect the customers via tunnel (for example, from the mobile devices. In some countries cellular providers block VoIP traffic and this helps to overcome this)
• A new option - per-node call limit has been added. This allows you to limit the maximum call amount passing through the node in order to not to have voice quality degradation in period of system overload. This combines well with cluster mode of operation, when customers operate on several IP addresses of your cluster. In case when one of the nodes is overloaded, customers get indication 'Switching equipment congestion' and switch to other node of your cluster.

Our company's product portfolio has been extended with Easy Gentoo - a free Linux distribution, that has been developed on the basis of Gentoo.

Distribution is a 100% authentic (without modifications) Gentoo system, that is installed by Debian installer.

Distribution will be basic for the Linux-versions of the Smartswitch product.

The reason for building our own distribution is that none of the existing ones completely fits our needs.