I am working with two very different implementation projects this winter. One is about using Asterisk in large call center environments. Another has a goal of building a SIP network for 15.000 phones in a university.

Asterisk for large call centers - full control

The call center platform is focused on a lot of PBX functionality. Every agent stays connected to a conference session, where customers are connected and disconnected, recording is enabled and disabled, dtmf is used to control various services and both AGI and AMI (the manager interface) are both being used heavily to integrate with a controlling application. Asterisk is in full control of each and every call, all the time, but the application controls the flow of the calls. The dialplan is very small for a large-scale Asterisk installation.

Large scale SIP networks - Asterisk on the edge, providing services

For the university, scaling is important. It’s a plain SIP network, with two computer centers in different buildings. All accounts are managed by LDAP, there’s no account defined locally in the VoIP platform. SIP proxys (Kamailio) rule this network and DNS is used for failover. Many services like call transfer, three-party conference calls and call forwarding will be handled by the phones. Asterisk serves as gateways to the old world, Nortel systems, and feature servers for IVR, voicemail and switchboard. No single server is in control of any call.

The power of open standards and open source

Two completely different designs, both made possible by Open Source and Open Standards. Both of them needs to scale. Both of them needs failover, redundancy and stability. And in both cases, we’re replacing expensive legacy telecom equipment with new platforms that will cost less to operate, that has a higher degree of interoperability and much more functionality than the previous solutions. Open telephony wins.

Kamailio - the successor to OpenSER - has been released in a new version, 3.0. This version is based on the merge between SER, SIP Express Router, and OpenSER code bases and developer teams. It’s a SIP server that you can run as a SIP proxy, a session border controller, a SIP load balancer or SIP application server. In version 3.0 you have the power of all the modules and applications in OpenSER and the raw strength of the SER core in one product. Go check it out today!

My friend and co-teacher Daniel-Constatin Mierla has written a series of articles on the soon-to-be released Kamailio 3.0 that highlights new features. The main thing is that this new version of the old OpenSER SIP server is merged with the original SER, as part of the SIP-router project. This gives Kamailio a large set of new features, most notably the rewritten kernel that has more stability, scalability and performance than the old OpenSER. Please visit Miconda’s blogg and learn about all the new features!

Best of New in Kamailio 3.0.0 - #1: include fileI’m starting a series of posts, to highlight the best new features in Kamailio (OpenSER) 3.0.0, of course, from my point of view, hoping to cover most of them before full 3.0.0 is out (RC3 was done yesterday).

During 2008, I worked with my Portuguese business partner Wavecom to build a large installation of Asterisk and OpenSER in Portugal. The network is now running 300 servers, 250 of them running Asterisk, FreePBX and OpenSER. It’s handling 200 connections to legacy PBXs of all kinds and over 10.000 phone numbers. During 2009 we’ve developed a failover solution, to make sure that every FreePBX server always stays up, regardless of the state of the hardware.

We’ve migrated 34 Universities to use Open Realtime Communication with Open Source. They are now migrating their user base from old PBX phones to modern VoIP phones.

Want to learn more?
Come to Amoocom in Germany, May 4-5, 2009 and listen to my talk!   And if you need help with similar large-size projects, contact me on info@edvina.net!