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Kamailio Development Training, Barcelona, Feb 10-11, 2011

January 5th, 2011 Posted in News

Next Kamailio Development Training takes place in Barcelona, Spain, during February 10-11, 2011.

The coordinator of the training is Daniel-Constantin Mierla. The goal of the event is to teach how to write your own code for Kamailio SIP Server. Note that it is not a training for VoIP administrators looking to learn how to configure and operate Kamailio-based SIP-VoIP platforms.

During the two days in the class, following topics will be approached:

  • internal architecture
  • SIP parser
  • memory manager
  • locking manager
  • database API
  • config file language interpreter
  • RPC interface
  • pseudo-variables and transformations framework
  • module interface – write your own extensions in C as modules
  • documentation docbook format

The price per attendee is 160 Euro.

Number of seats is limited and access will be granted in first come first served fashion. Registrations or requests for more details must be done via email at registration [at] kamailio.org .

For people that happens to be in the area but not interested in programming C and/or not participating to the class, there will be a Social Networking Event during the evening of Feb 10, where we meet for food, drinks and discussions about SIP, VoIP and more.

Participation to this event is open and free for anybody, very likely to be a dinner or a pub session where every participant pays for himself/herself. Registration is still required so we know how many seats to reserve and send you the location of the event. Among people you will meet there are going to be Daniel-Constantin Mierla, Jesus Rodriguez and Inaki Baz-Castillo.

Kamailio Advanced Training, Jan 24-26, 2011, Irvine, CA, USA

November 11th, 2010 Posted in News

Next US and North America edition of Kamailio Advanced Training will take place in Irvine, CA, USA, Jan 24-26, 2011.

Last stable series of Kamailio SIP Server, the 3.1.x (Oct 06, 2010, see release notes), continues the work done within SIP-Router.org project. Among brand new features in v3.1.0, starting with the previous 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 in collaboration with Flowroute and will be taught by Daniel-Constantin Mierla, co-founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

http://www.asipto.com/index.php/kamailio-advanced-training-usa/

Siremis v2.0.0 Released

November 1st, 2010 Posted in News

Siremis v2.0.0 is out – the web management interface for Kamailio (Openser) v3.1.0 and SIP Express Router (SER).

This is a major release, with countless improvements and new features since v1.x series, among them:

  • major re-factoring of web interface
  • better accessibility
  • simplified menu structure
  • completely new look
  • dashboard with the map of all available tools
  • developed on top of Cubi and PHPOpenBiz v2.4 frameworks
  • web installation wizard
  • added new modules: xcap, dialog, new lcr
  • usage of separate database for siremis itself
  • management of users that can login to siremis
  • management of menu can be done from web interface
  • building Apache conf and htaccess file can be done by Makefie
  • charts to monitor location transport layers

Step by step installation tutorial, screenshots and demo are available on the web at:

Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good oportunity to learn about Siremis itself, next locations are:

Kamailio Advanced Training, Nov 22-25, Berlin

October 19th, 2010 Posted in News

Next Kamailio Advanced Training will take place in Berlin, Germany, Nov 22-25, 2010.

Last stable series is 3.1.x (Oct 06, 2010, see release notes), continues the work done within SIP-Router.org project. Among brand new features in v3.1.0, starting with the previous 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, co-founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

Kamailio v3.1.0 Released

October 6th, 2010 Posted in News

Kamailio (OpenSER) v3.1.0 is out – major release with impressing number of new features and improvements.

On October 06, 2010, Kamailio (OpenSER) 3.1.0 has been released – this release is a result of more than 8 months of development from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Backed up by a solid development group, we are proud to announce that this release brings a large set of features,  many for first time on the SIP server market, such as asynchronous TLS, UDP raw sockets, embedded HTTP and XCAP servers, embedded Lua, configuration file debugger. All together, there are over 15 new modules and countless improvements to old components.

Since last major release (version 3.0.0, which was out in January 10, 2010),  the two SIP servers, Kamailio and SER, are practically the same application, the name making the difference regarding the database structure and the extensions used for certain features, such as user database based authentication or location service. Therefore another development direction was towards smooth integration of Kamailio and SER extensions, previously duplicated modules such as auth, sl, ratelimit or sms were merged during this development cycle.

Continue reading the release notes at:

http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

E-Learning class to start on Oct 4, 2010

September 10th, 2010 Posted in News

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.

SIP Router Masterclass, Nov 8-12, 2010, Berlin, Germany

September 6th, 2010 Posted in News

Next SIP Router Masterclass will take place in Berlin, Germany, November 8-12, 2010.

Click to download course description brochure

Brochure

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (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 (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. Starting with version 3.0.0 (highlights), you can use mixed features and modules from Kamailio (OpenSER) and SIP Express Router (SER) in the same configuration file.

The course targets system administrators and people that act in large VoIP/telephony network services, as well as integrators of VoIP, instant messaging and presence with web 2.0 or similar technologies.

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.

Presentation at ClueCon 2010

July 23rd, 2010 Posted in News Tags: , , , ,

Daniel-Constantin Mierla, Asipto’s representative (co-founder and member of management board of Kamailio (OpenSER)), is going to present Kamailio SIP server at ClueCon, Chicago, USA, Aug 3-5, 2010.

The talk will focus on the last stable version and what is new in development branch as well as usage of Kamailio together with FreeSwitch.

If you want to meet during the event and learn more about our solutions for IP Telephony and Unified Communication, please contact us via email or web form at:

http://www.asipto.com/index.php/contact-us/

Asterisk SIP Masterclass Summer 2010

June 5th, 2010 Posted in News Tags: , ,

Asterisk SIP Masterclass is professional training program dedicated to Asterisk and integration with SIP router (Kamailio). This class focuses on building a carrier-class scalable network architecture with Asterisk and Kamailio.

The course will be taught by two teachers that have all the insights you need to learn the details of Asterisk and Kamailio (OpenSER):

  • Olle E. Johansson – Asterisk SIP Developer and Kamailio (OpenSER) contributor
  • Daniel-Constatin Mierla – Kamailio (OpenSER) Developer and founder

The class is held in Malaga, Spain, July 5-9, 2010.

Register now – seats are limited! It is more than a learning opportunity, a great chance for networking and business opportunities, a place to get in touch with latest VoIP technologies and professionals world-wide.

For more details and registration:
http://www.avanzada7.com/en/SIP_Masterclass