Kamailio Practical Workshop is a new concept of knowledge transfer event offered by Asipto, coordinated by one co-founder of Kamailio, Daniel-Constantin Mierla, a person with many years of expertise and hands on experience with VoIP and Rich Communication Services. It combines consultancy and training services, targeting to offer practical results that are needed by each participant.

The structure of the workshop:

• each participant comes with own set of requirements to build Kamailio-based platforms during the workshop
• the coordinator will identify the topics of common interest and have dedicated sessions for them
• participants can group together to work and achieve the building of VoIP systems they need
• the coordinator will be there to guide, help to build and troubleshoot during the workshop

Date and Location:

• September 10-12, 2012
• Alkmaar, The Netherlands (45 min from Amsterdam airport)

More details about the event and registration process at:

• http://www.asipto.com/index.php/kamailio-practical-workshop/

Next US and North America edition of Kamailio Advanced Training will take place in Seattle, WA, USA, during September 24-26, 2012.

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 process at:

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

Daniel-Constantin Mierla of Asipto, co-founder and member of management board for Kamailio project, will provide one hour workshop during LinuxTag Conference & Exhibition, Berlin, Germany. Scheduled on Thursday, May 24, 2012, the workshop will focus on how to build yourself a secure unified communication platform, using Kamailio in server side and Jitsi as application in client side.

Asipto’s portfolio include a set of solutions for securing and scaling IP telephony services, with options to add rich communication services such as video, instant messaging, presence or desktop sharing.

You can read more about the workshop at:

• http://www.linuxtag.org/2012/en/program/project-workshops/details.html?talkid=705

As you get at LinuxTag show, just drop by at Kamailio project booth to have a chat, you can meet people from Asipto team, five of the management team members, along with other developers, community members as well as the friends from SIP Express Media Server (SEMS) project. The booth is located in Halle 7.2b, Stand 278.

ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry.

The event took place at the House of Commons Members Dining Room, Palace of Westminster, London, on 21st of March 2012, hosted by Dr Julian Huppert MP, Vice Chair of the Parliamentary Internet Communications and Technology Forum.

With main focus on awarding IP Telephony businesses in UK, this year they introduced a new category, “Members’ Pick Award”, to endorse something or someone that has provided real value to VoIP Industry. Open Source VoIP Projects as a group was introduced in this category, made it do the final and ultimately won the category.

Daniel-Constantin Mierla, of Asipto, co-founder and member of management board of Kamailio SIP Server project attended the event and was selected to pick up the award.

The members of ITSPA acknowledged the major role of open source VoIP projects for their businesses, many of them would have not existed without these projects. Awarding Open Source VoIP Projects as a group was decided because most of the deployments combine several projects to build a complete IP telephony platform. It is a common practice to mix applications such as Kamailio/SER, Asterisk or FreeSwitch to build large VoIP systems and provide a broader range of services.

It is yet another confirmation of reliability and quality solutions provided by Open Source environment for real time communications.

This is also an opportunity to send best wishes and regards to all the people behind Open Source VoIP Projects, developers or community members, that dedicate work and time to develop and improve the quality of the applications and act in the true spirit of Open Source: freedom and fairness!

Fosdem 2012 included a DevRoom for Open Source Telephony, Daniel-Constantin Mierla of Asipto participated and presented “Secure SIP Communication with Kamailio”.

First part focused on an overview of the project, history and latest new features, then continued to present the tools offered by Kamailio to achieve strong security in your VoIP deployments.

You can find the slides (pdf) at:

• http://asipto.com/u/54

If you have questions related about SIP security or look for consultancy in this area, our experienced team can help you, feel free to contact us.

Siremis v3.2.0 is out – the web management interface for Kamailio SIP Server (former Openser) and SIP Express Router (SER). This is a major release, compatible with Kamailio v3.2.x.

This release brings a large set of new features. Among them:

• SQL-based CDR rating engine for billing purposes
• stored procedure to compute the costs of calls
• Management of billing rates
• longest prefix rate selections
• rating rules can be grouped to allow many sets of values
• time unit is configurable per rating rule
• Management of remote registration records (uacreg table)
• Managment of mtree module (mtree and mtrees tables)
• Management of dialog variables table
• Update of LCR and SIP Trace views for compatibility with Kamailio 3.2.x
• Tools to generate new database table views in a wizard fashion
• create new views to database table with a command line tool in 5 steps
• Charts drawing statistics of accounting records
• graphics to show the evolution of accounting records during the past hours
• graphics to show the types of INVITEs (call setup) during the past hours
• Tables presenting summary of accounting records
• count the number of INVITEs and BYEs in the past hours
• present the top activity of accounting records – e.g., top 5 caller and callee
• more can be added from configuration file
• More SIP server activity charts (e.g., SIP requests traffic load)
• e.g., default chart presents how many requests are received in intervals of 10 minutes
• Buttons to switch to command pannel to reload Dispatcher or PDT records in SIP server cache
• once new records are added, in two clicks they get in the cache of SIP server
• Views for managing global black lists table
• Many improvements to user interface
• selection of local domain is done via select box or picker form (e.g., in aliases, user preferences, pdt, …)
• selection of local username is done via picker form (e.g., user black lists, user preferences, aliases, …)
• group names can be set in config file and selected from a list box
• many static values are given as option to select from a list box (e.g., dispatcher flags, lcr options)
• More targets in Makefile to make administration easier

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

• Installation Tutorial
• Screenshots
• Web Demo for Siremis 2.0 – v3.2.0 is built on same core framework, just new features were added (thus more options in the menu)

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

• Kamailio Advanced Training Presentation Page