From now on Smartswitch is available in native format for 64-bit platforms.

This allows to overcome restrictions of 32-bit mode of operation:

• each process can access 4G of RAM maximum (because of 32-bit address space);
• whole OS can use up to 4G of RAM (even in case if 8G are installed). The only workaround is to use PAE (Physical Address Extension) mode. But this mode in turn puts other restrictions - not all drivers can be used and kernel has to be built monolithic (no modules).

64-bit OS solves these problems and enables 64-bit CPU registers, which should boost performance.

A Smartswitch build for 64-bit platform consists of the same components as the 32-bit one. Therefore you should prefer using it in case if your platform supports 64-bit.

We are glad to introduce to you a new 6.3 release of our telephony platform - Smartswitch.

In this release you will see new features as well as bug fixes. As usual, here is a brief list of changes:

• Implemented call hangup on RTP timeout for cases when media goes through the Media Proxy;
• New SIP options added: "insecure port" and "insecure INVITE". These options are used to control the SIP user authentication;
• implemented Call Handler launch from a Google Chrome web browser;
• implemented sorting (ascending or descending) by call start time or call end time in Call Detail Report;
• CDR storage scheme has been reviewed. New scheme saves 30% of disk space compared with a previous one;
• system's performance related to peer call limits application has been greatly improved. New scheme allows to decrease CPU consumption to 50% during the flow of 500 calls and ASR 30%;
• implemented feature of displaying a price in the active calls tab. You can choose whether to display per-minute price or to display a whole call price for the current duration;
• implemented "constant SSRC" option for SIP and H323 channels. This option controls SSRC change in RTP stream after media has been started processing by the Media Proxy. This option is needed for compatibility with some gateways (to be more specific, Huawei);
• statistics calculation for Media Proxy has been implemented. Currently this includes only sent/received packets counters;
• we have moved to HTTPS web interface, a possibility to change SSL certificate has been added to web interface;
• a Loss Controller has been added;
• billing performance has been improved by the factor of 2;
• implemented possibility to change billing pack and routing configuration for the user of the Calling Cards scheme depending on the DID called;
• fixed issue of displaying of the invoiced sum in the Financial Report.

A Demo for the Smartswitch system has been updated to recently released 6.2.

Enter and explore!

We are glad to announce a new 6.2 release of Smartswitch.

In this release you will see new features as well as bug fixes. As usual, here is a brief list of changes:

• Fixed bug in openh323 library.

  Due to this bug under certain circumstances traffic from H323 originators has been switched to terminators (both SIP and H323) via codec transcoder. However traffic could easy go without being converted (for instance, we got g729->ulaw->g729 call). This caused unnecessary CPU load and voice quality degradation (caused by double transcoding). This problem has been revealed due to Media Format Report, which was introduced in release 6.1. Problem is relevant for the users of all previous releases.

• Fixed bug in SIP channel, which caused denial of service under load (above 500 simultaneous calls) in certain circumstances, connected to T38 fax transmission. Problem is relevant for the users of all previous releases, which use system under high load.

• Removed possibility to refill user using his own refill PIN code.

  Because after this user has been deleted from the system. Problem is relevant for users of Calling Cards, which use refill from other account (not voucher) feature.

• Fixed logging of user removal.

  When removing subscriber's accounts this operation hasn't been logged and user's data hasn't been saved in changelog. After this subscriber could be restored only after appealing to our tech support service. Problem is relevant for users of Calling Cards solution.

• Fixed call limit application for subscribers.

  There have been some circumstances, when call limit has been applied incorrectly. Problem is relevant for users of Calling Cards and Service Provicer solutions.

• Fixed memory leak in web interface which happens during active calls monitoring.

  Problem is relevant for the users of version 6.0 and above.

• Fixed CDR export to file, that was broken in STANDARD edition. Relevamt for users of STANDARD edition.
• Fixed issue of role permissions delimitation while exporting the subscribers to a file.

  Agent/dealer could see not only the subscribers he own but all the others too in the resulting file, however web-interface displayed everything correctly. Problem is relevant for the users of release 6.1

• Inter-operation with clusters has been enhanced.

  Added possibility to interconnect with clusters that implement load balancing using the SIP message "302 Moved Temporarely" and to correctly bill the outbound calls. This is the scheme that MVTS clusters use.

• New tabs for subscriber's web-portal have been added:
  • Check Price - subscriber can enter destination number and get its price;
  • Callback - subscriber can trigger callback within web-browser.

• Added option to control CDR details displayed in subscriber's web-portal: when needed you can hide redundant information;

• A new tab has been implemented for administrator web-portal - SIP Registry.

  This tab contains outbound registrations that are active at the moment and status for each of them, as well as the time of last message sent.

• New options for SIP session timers control have been added onto administrator web-portal.

  These options have been added to global section as well as independently for each dial-peer. Session timers serve to hangup calls in case if keep-alive message exchange has been dropped. The latter can happen for example in case if internet connection has been dropped.

• A сumulative financial and summary reports for companies and subscribers have been implemented.

  This enables you to see the complete picture on the financial situation with a single look. These reports work in team-up with subscriber filter. Therefore, you can filter your subscribers using some criteria first (for example, zero balance), and then look at the cumulative financial report for the filtered subscribers.

• A filter by balance has been added into administrator web-portal into the list of subscriber's filters;
• An Invoice Generator has been added into administrator's web portal.

  Invoice Generator enables you to automatically generate total amount of invoice and to compose invoice according to template provided by you. Resulting document can be either in .pdf or .html format and can contain graphical elements inside, for example, company's logo.

• Added feature to change tariff plan and routing depending on the DID number called.

  This feature is relevant to users of Calling Cards.

• A logic of playing back an "enter PIN" prompt has been changed.

  Previously the same file has been used both for authentication prompt and for refill prompt. From now on you can define separate files for each function. This feature is relevant to users of Calling Cards;

• A billing/routing performance has been improved.

  Due to new caching strategy introduced you can get performance increased up to 30% depending on configuration and traffic.

• We have moved from byte/sec to bit/sec on the graphics of interface load, because latter is more usual to users;
• Navigator menu (the one on the left) has been enhanced in all portals;
• Other minor enhancements and fixes;

We are very grateful to all system's users (both commercial and non-commercial) for bug reports and interesting ideas how to enhance functionality. We very appreciate your help and support!

We are glad to announce that Smartswitch system has been successfully connected to MVTS cluster by one of our customers for VoIP traffic exchange.

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.