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 be present at Call Center World congress in Berlin, February 8-11, 2010. If happens for you to be around and want to meet, feel free to contact us.

The event gathers over 250 exhibitors from around the globe, mainly focusing on help desk and support solutions, integrating SIP/IP and TMD for call centers. Along with the exhibition, there are workshops and the conference.

Asipto’s Load balancing solutions can be used to scale call center capacity in a transparent and flexible manner.

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.

Daniel-Constantin Mierla, head of SIP Solutions, will give a presentation at 3G+ and LTE Network Planning and Optimisation Forum, taking place in Berlin, Hotel Palace, January 20-22, 2010.

The speech will focus on SIP Applications for 3G+ Networks:

• Demands and challenges of SIP for quality communication service
• SIP for Voice and beyond – build scalable new services in 3G+
• Quality of service versus quality of experience
• SIP as bearer for QoS metrics and monitoring data
• What attractive services can be built using SIP
• How SIP can be used to drive the maintenance of networks 3G+ Optimisation (e.g., monitoring nodes, capacity, optimising routes)

If you are in Berlin during this period and want to meet, please contact us.

On Jan 18, 2010 Asipto announced new major version of SIREMIS Web Management Interface for Kamailio (former OpenSER) and SIP Router is available as v1.0.0:

http://siremis.asipto.com/

Among new features there are new monitoring charts, sip trace and presence services support. This release is fully compatible with new Kamailio (OpenSER) v3.0.0 and SIP-Router.org

Online demo (user=admin, password=admin) was updated to run Siremis v1.0.0

Daniel-Constantin Mierla, co-founder Kamailio (OpenSER), announced availability of version 3.0.0 of the open source SIP server.

It is a major release of Kamailio (OpenSER), following ten months of development and heavy testing. Represents a special release, being based on SIP Router project, therefore enables admins to blend Kamailio (OpenSER) and SIP Express Router (SER) core features and modules in same configuration file.

Given the above, this version brings out an impressive number of new features and improvements, read the release notes at:

http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes

Asipto will release soon a new version of SIREMIS, the web administration interface for Kamailio, to work out of the box with v3.0.0.

Daniel-Constantin Mierla, co-founder Kamailio, announced the availability of Kamailio (OpenSER) v3.0.0-RC3 – last release candidate before full 3.0.0 major version – the tarball with sources of Kamailio 3.0.0 RC3 is available at:

http://www.kamailio.org/pub/kamailio/3.0.0-rc3/src/kamailio-3.0.0-rc3_src.tar.gz

Lot of fixes to packaging, code and documentation have been committed since RC1, Kamailio 3.0.0-RC3 becoming ready for pre-production phase.

If you like to work with source, then use the tutorial guiding the installation of Kamailio 3.0 branch from GIT repository:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git

Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version:

http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x

http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0

Feel free to contribute to wiki pages, helping to build migration tutorials.

This is the last RC before full 3.0.0 release. Please report any issue you find to sr-dev at lists.sip-router.org.

On short notice, Asipto‘s representative, Daniel-Constantin Mierla, will present Understanding SIP/VoIP Architecture Design at SIP and IMS for Next Generation Telecoms 2009, September 23-25, Berlin, Germany.

The presentation is held on Sep 23, 11:30, focusing on:

• Optimising next generation SIP networks
• Planning and implementing new SIP functionalities
• Using SIP for prepaid systems and internet telephony platforms
• Integrating load balancing and session border control

If you are in Berlin during the event and want to meet, contact us.

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.

Click here for course details and registration.