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.
Hey! About us: 1. telephony systems for business development and implementation 2. systems, that are based on Asterisk 3. systems creation for call centers, with automatic customer calling (vicidial) 4. graphical user Daha Fazla
Bu iş için 9 freelancer ortalamada $160 teklif veriyor
I have 8 years of experience in voip solutions and deployment. I can provide better option with easy to manage gui.. I can start work right now.