I have been working on a project with asterisk and Kamailio. I would prefer using Kamailio because i have personally met with the developers and it has more active users and rapid developments. The developers are also very friendly and helpful. And well OpenSER is not gone, the name is changed to Kamailio I guess. It had a fork, but now they have merged together.
NOTE: this is a practical, hands-on book based around a fictitious case study VoIP Provider that you will build on a development server using OpenSIPS 1.6. The case study grows chapter by chapter, from installing your local development server, right up to the finished VoIP provider. This book is for readers who want to understand how to build a SIP provider from scratch using OpenSIPS. It is suitable for VoIP providers, large enterprises, and universities. Telephony and Linux experience will be helpful but is not essential. Readers need not have prior knowledge of OpenSIPS. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS.
The folks of OpenSIPS are similar, docs from one can help in the other. The opensips mailing list is monitored by one of the main developers. He is even in the IRC chat in the mornings. The docs are kept current on the opensips webpage. They like to change modules a bit, so really watch your versions. The commercial PDF “Building Telephony Systems with OpenSIPS 1.6″ is excellent. Yum is nice for the dependencies, but I would use a compile for Opensips. Most of the docs are Debian specific. I love Debian, but our clients love Centos. I have some Centos Opensips compile docs if needed.
There are a few GUI’s, but I prefer Opensips-cp. To put opensips-cp on a remote server, you need the xmlrpc module loaded on opensips. This works in Debian but fails on Centos (64 bit ONLY).