Hello everyone, I hope that you are passing a good time,
well I want to do a small lab : integration of Asterisk and OpenSIPS for Load balance Voip Traffic calls purposes, I have already installed Opensips v2.4 in my debian machine and Asterisk v 13,
but i really dont know how to start to configure and Load balance the calls and route them,
I will appreciate your help or suggestions or anything,
I have been searching for like 2 weeks in the web, I didn’t find something interesting and detailed about the OpenSIPS and Asterisk integration( Only installation etc …), the documentation also is limited, I can say that kamailio is providing more resources than OpenSIPS.
This is exactly how it works in Kamailio because OpenSIPs is the fork of OpenSER and when they forked OpenSER it was a case of how they wanted to move forward so it became Kamailio (OperSER) and OpenSIPs (the forked project of OpenSER). So they both follow the exact same logic, syntax and structure. In fact they share a lot of the same core modules.
I want to configure Kamailio server so that traffic will be forwarded to other 2 asterisk servers equally
i have already read about the dispatcher module
i think there is a list i should edit to add the 2 SIP addresses,
for the dispatcher list, should i create one ? or it’s already there ?
Thanks in advance
I know openSIPS and Kamilio got started from same base project. For us when we are evaluating both, we felt openSIPS has better documentation & tutorials, which will be helpful for someone who is new to load balancing VoIP solutions.