Posts Tagged ‘training’

Kamailio E-Learning Class, Oct 4, 2010

A new E-Learning class about SIP Router Configuration File is due to start on Oct 4, 2010. Registrations are accepted up to Oct 2, 2010.

The class duration is three months and gives the opportunity to learn the structure of configuration file and how to write it properly. Lessons are applied to Kamailio (OpenSER) and SIP Router SIP servers, touching VoIP security and scalability, at a fee of just several consultancy hours.

Kamailio (former OpenSER), now at release v3.0.0, is an open source SIP server, awarded Best of Open Source Networking Software 2009 by InfoWorld magazine, used world wide in VoIP platforms servicing millions of active subscribers and routing billions of call minutes per month.

Target attendees:

  • VoIP administrators willing to learn and have a support channel for SIP Router configuration file
  • System and network administrators willing to enhance their portfolio with VoIP knowledge

Certificate can be achieved at the end of the class.

Please use the contact form for registration details.

Edited: September 10th, 2010

Kamailio SIP Masterclass, March 22-26, 2010, Berlin, Germany

Next SIP Router Masterclass will be held  March 22-26, 2010 in Berlin, Germany.

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

End of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio (OpenSER) is now at version 3.0.0 (released on January 11, 2010), being based on SIP-Router.org project.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Click here for course details and registration.

Kamailio (OpenSER) and SER are the leading open source SIP servers, routing billions of minutes and handling millions of active VoIP users each month. Well known for its stability and flexibility the SIP Express Router (SER) family of SIP servers is continuously increasing the adoption on the market. With the launch of SIP-Router.org project, the SIP server secured a reliable development team, backed up by strong business community.

Edited: February 11th, 2010

Kamailio SIP Masterclass, November 9-13, 2009, Berlin, Germany

Next SIP Router Masterclass will be held  November 9-13, 2009 in Berlin, Germany.

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

By end of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio 3.0 and SER 3.0 (to be released soon) become compatible in terms of configuration file and extensions.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Click here for course details and registration.

Edited: September 1st, 2009