News for September 2010

Siremis v1.0.1 Released

A new version of SIREMIS Web Management Interface for Kamailio (former OpenSER) and SIP Router is available as v1.0.1.

Siremis enables straightforward management of subscriber profiles, least cost routing and load balancing rules, communication at runtime with the SIP server, displays monitoring charts. You find detailed list of feature at:

http://siremis.asipto.com

v1.0.1 is a minor release, fixing issues discovered since releasing last version:

  • replaced deprecated PHP functions such as split() and ereg()
  • added missing column in LCR gateways
  • set default page to home.php instead of login.php
  • set default path to XMLRPC commands in configuration

Download and installation steps:

http://siremis.asipto.com/install/

Some screenshots specific for this version:
http://www.asipto.com/gallery/v/siremis/siremis_22.png.html
http://www.asipto.com/gallery/v/siremis/siremis_23.png.html
http://www.asipto.com/gallery/v/siremis/siremis_24.png.html

More screenshots:

http://www.asipto.com/gallery/v/siremis/

Demo site (it works on a database with random data, username: admin, password: admin):

http://siremis.asipto.com/demo/

Edited: September 14th, 2010

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

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

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.

Edited: September 8th, 2010