Integrating Asterisk And OpenSIPS In Realtime : Tutorial

Integrating Asterisk And OpenSIPS

We have been working with Asterisk and OpenSIPS for a long time (Even when OpenSIPS was OpenSER!). But we had to hammer out solutions as they fit and there were no methodology to it. Various engineers had their notes and were collected at one place only to be pulled out when the need arises to bring these two Open Source Telephony servers together.
SIP server like OpenSIPS usually provides registration and call handling, either by routing, forwarding or direct connections. The Asterisk on the other hand is a full blown telephony server, providing IVR, Voicemail, conferencing and announcements in addition to call handling, So by bringing these two servers (services) together, one will be able to provide full blown telephony solution integrating SIP, and Asterisk services.
So without saying that our method(s) are messy, I am going to point you to a very nicely written (Also very nicely thought out) article on how to integrated Asterisk and OpenSIPS.
Bogdan Andrei Iancu, a core OpenSIPS developer has written the tutorial. Go read it, Print it and save it! It is very good!
PS The tutorial currently addresses Asterisk 1.4 or 1.6 and OpenSIPS 1.5x. Once the OpenSIPS 1.6 is released the documentation will be updated appropriately.
Realtime OpenSIPS - Asterisk Integration