I have a FreeBSD server with asterisk16 installed via packages. I want a MINIMAL working asterisk SIP configuration to accomplish these things:
1. Three softphone extensions that can call each other. Two on desktops behind internet firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself.
2. A publicly available extension, like me@[login to view URL], that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me).
3. Everything secure: No unauthorized access.
4. Be able to switch between RTP and SRTP by changing the configuration.
5. If possible: no asterisk database connection, only flat files. If not possible, explain why.
6. Asterisk comes up and runs as cleanly as possible: no unexplained error/warning messages.
7. Use the defaults that come with asterisk installation where possible. No modifications that will hinder a "pkg upgrade".
I cannot grant access to the server, so I'll have to just put your files on the server and try them myself. A screen-share may be possible for troubleshooting.
If something other than asterisk16 is better, talk me into it. (Asterisk13? FreePBX? Kamailio? ) Better means simpler to configure and understand. A web-based configuration tool could be useful, but not necessary. I don't mind editing files on the server. Any proposed solution must run on FreeBSD.