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 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:

• Installation Tutorial
• Screenshots
• Web Demo

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 for Carrier Services, November 22-25, 2010, Berlin, Germany
• Kamailio Advanced Training for Carrier Services, January 24-26, 2010, Irvine, CA, USA

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:

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

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

Click to download course description 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.

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/

VoIPToday has published an interview with Daniel-Constantin Mierla, co-founder and core developer of Kamailio (OpenSER) project, executive member of Asipto. Read it at:

• VoIPToday Kamailio Interview

Kamailio (former OpenSER) SIP server is a winner of InfoWorld Awards: Best Open Source Networking Software 2009, confirming one more time it is the leader of Open Source SIP servers out there.

Last stable series is 3.0.x (Jan 11, 2010, see release notes), the first based on SIP-Router.org project. Among brand new features, it lets you 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.

Next class Kamailio Advanced for Carriers, organized by Asipto in collaboration with IPCom Network and NAP of the Americas, focuses on the features that allow you to build a carrier SIP infrastructure:

• least cost routing
• load balancing
• traffic dispatching
• DID routing
• prefix based routing
• SIP trunks and peering
• traffic shaping
• topology hiding
• flood detection
• enum
• authentication and authorization
• accounting
• redundancy
• high availability and failover

Location: Miami, USA
Date: June 21-23, 2010
Trainer: Daniel-Constantin Mierla
Registration price starting at: 1795 USD

More details and registration at:

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