Hi! I just came in possession of a Cisco 7942 IP phone and I am eager to set it up. I did some research and discovered how to flash it with SIP firmware as opposed to the SCCP firmware. As for the actual SIP server, I discovered Kamailio and decided on this one as it is in the main package list for OpenBSD which is the platform I mainly want to develop this on. I also discovered Siremis as a GUI for Kamailio, but it seems development has been lacking. Does anyone have any tips or tricks or advice on setting up a SIP server? Anything weird I need to look out for? Any challenges you guys specifically faced when setting up a SIP server? Thank you for any tips and advice!
Kamalio is far from being user-friendly and is usually used by service providers as a proxy/session border controller rather than a PBX. Unless you are planning on using it as a proxy there will be no real benefit over more user-centered PBXs. Since you were trying to find a GUI solution for Kamalio, my guess is that PBX is what you actually looking for.
I was actually able to get Kamailio up and running on a VM and got a successful ring between two other VMs. A GUI isn’t terribly important, just a little convenience to have. How does Asterisk compare to Kamailio? I’ve read that while Asterisk is more user friendly, Kamailio is more stable and can support more lines. However, I have also read you can run and load balance Asterisk on Kamailio
Asterisk is way more user-friendly and a good starting point. It all really depends on what your goal is. It is very stable if compiled correctly.
Kamalio is a great robust load balancer and is widely used for this exact purpose. Originally it didn’t even have the RTP module and was handling SIP signaling only.
The biggest hit in terms of productivity comes from handling RTP traffic and transcoding. That’s pretty much why one Kamalio can service multiple Asterisks.