Kamailio Advanced Training
November 13-15, 2017, in Berlin, Germany
Click here for more details!
Learn how to build RTC services with Kamailio!
Tuesday, November 29, 2011
Thursday, November 24, 2011
Next Kamailio SIP Server Advanced Training will take place in Berlin, Germany, Dec 5-8, 2011.
Last Kamailio stable series is 3.2.x (Oct 18, 2011, see release notes), continues the work done within SIP-Router.org project. Offering a big lot of brand new features in v3.2.0, starting with an older major version, 3.0.0, you can run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.
The class is organized by Asipto and will be taught by Daniel-Constantin Mierla, founder and core developer of Kamailio SIP Server project.
Read more details about the class and registration at:
Wednesday, November 23, 2011
Siremis v2.1.0 has been released – this is an update to previous release v2.0.0, bringing several enhancements and new web pages to manage PUA and RLS. It is still compatible with Kamailio v3.1.x, the last of this kind, next one to be out in the near future will be compatible with Kamailio v3.2.x.
You can find the news about this release, including links to download, screenshots and demos, at:
Alternative download site (tarball or git pull) is from sourceforge project:
Siremis v2.1.0 is working for most of the components with Kamailio 3.2.0, just the few that changed the database structure may not be fully functional (e.g., the modules with tables that have new columns, see http://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0#sql_commands).
Saturday, November 19, 2011
The profile data document format is usually specific per user agent, such documents have to be built and added to presentity table by the admin or a third party application.
Read more about SIP user agent configuration framework in RFC6080:
Tuesday, November 15, 2011
The module reuse existing xhttp module, therefore it has no external library dependencies and the processing rate matches the performances of processing SIP requests.
You can read more about this module, see an example of how to use, at: