Every year the spring comes with CeBIT exhibition in Hannover, Germany, the biggest show of IT industry. Just a search on VoIP keyword revealed 136 exhibitors. It looks very promising.
Asipto representatives will be present at the show, starting with Wednesday, March 3. If you want to meet, feel free to contact us and set an appointment via email or web form at:
Next SIP Router Masterclass will take place March 22-26, 2010 in Berlin, Germany.
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.
Asipto and Quobis work together to market and support SIP solutions for telecom operators and governments in Latam and Iberian markets.
Quobis Networks and Asipto today announced a set of business collaboration agreements to market and support a series of solutions based on Session Initiated Protocol (SIP) for telecom operators and governments in LATAM and Iberian markets. Under this new model, customers will realize unprecedented flexibility through improved interoperability and manageability of the opensource solutions developed by Asipto.
Quobis will provide professional services in the phases of analysis, planning, design and certification for projects related with implementations of open source unified communications, where the company has a broad experience. During the last four years Quobis has worked in some of the most important projects in iberian markets.
Main solutions included in this new portfolio are unified communications (IP communication sessions with NAT traversal support, presence and instant messaging, rich telephony features and media services), load balancing (customizable load balancing rules, with safeguard module to secure the SIP internal network, offer high availability and failover), prepaid engine (with per-second precision billing and multiple charging plans) and number portability (Number Portability Server with high capacity of records for ported numbers and ability to process thousands of requests per second).
“This agreement reaffirms the commitment of our company to become the leading technical consultancy in Latam and Spain and contributes to complete our portfolio of services for telecom operators” states Iago Soto, Chief Marketing Officer of Quobis. Daniel-Constantin Mierla, co-founder of Kamailio (OpenSER) is confident that “this agreement makes possible to expand our developments to new markets addressing operators and companies through our new Spanish partner”
About Quobis
Quobis is an spanish company that helps leading companies turn challenging communication requirements into opportunities for growth using telecom emerging technologies like ITS, Smartgrids, Mobile WiMAX or open source VOIP. The team is formed by senior telecom engineers, with several years of professional experience working on consultancy, design, planning and project management within these technologies.
About Asipto
ASIPTO has a strong background in Kamailio (OpenSER), SIP and VoIP. Backed up by two co-founders of Kamailio (OpenSER) SIP Server Project, the knowledge of the team is built on the experiences in VoIP since beginning of 2002, with world wide production deployments and active development of Kamailio (OpenSER).
Daniel-Constantin Mierla will give a lightning talk about Kamailio (OpenSER) 3.0.0 on the main track at FOSDEM 2010, the annual meeting of free and open source developers.
It is already a tradition that folks around Kamailio (OpenSER) and SIP-Router.org projects meet in Brussels to plan the year. Last time we crafted the SIP Router integration plan, which seemed impossible to be doable in an usual release cycle, but it was with luck, 3.0.0 is already out there.
A new E-Learning class about SIP Router Configuration File is due to start on February 8, 2010. Registrations are accepted up to February 5, 2010.
The class duration is six 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:
Certificate can be achieved at the end of the class.
Please use the contact form for registration details.
Next SIP Router Masterclass will take place November 9-13, 2009 in Berlin, Germany.
Teachers:
Daniel-Constantin Mierla – co-founder of OpenSER/Kamailio 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.
InfoWorld awarded Kamailio the Best Open Source Networking Software 2009, acknowledging the wide spread, maturity and its large set of capabilities. Kamailio 3.0 is due to October 2009, the most scalable and feature rich release in its series of SIP server so far.
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.
Asterisk SIP Masterclass is professional training program dedicated to Asterisk and integration with SIP router (Kamailio/OpenSER). This class focuses on building a carrier-class scalable network architecture with Asterisk and Kamailio (OpenSER).
The course will be taught by two teachers that have all the insights you need to learn the details of Asterisk and Kamailio (OpenSER):
The class is held in Malaga, Spain, June 22-26, 2009.
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
E-Learning class on SIP Router Configuration File is due to start on June 3, 2009. Late registrations are accepted up to June 12, 2009.
The class duration is six months and gives the opportunity to learn how to write proper configuration files for SIP Router and Kamailio (OpenSER) SIP servers, how to take care of VoIP security and scale the service. Target attendees:
Please use the contact form for registration details.
Daniel-Constantin Mierla, Asipto, will present “SIP Server Routing Platform” at AmooCon 2009, May 4-5, Rostock, Germany. The talk will focus on SIP-Router.org Project (Kamailio (OpenSER) and SIP Express Router (SER)), benefits and tools to build new services using the configuration file.
Details at:
The third edition of ROSDEV – Romanian Open Source Development Meeting – takes place in Brasov, Romania, April 04, 2009.
The intiative was started by Elena-Ramona Modroiu, COO Asipto, back in 2006, targeting to build the event that gathers Romanian developers working with Open Source projects.
This year the conference goes international, with invited people from other countries. Asipto team will present its new SIREMIS project along with Kamailio (OpenSER) and SIP-Router.org projects.