I am very new to Asterisk, and I was hoping someone here would assist me. I am using the latest GitHub version (as of June 20th.)
I am interesting in streaming the audio (via Unicast) of each incoming SIP call. I found this blog post which seems to detail how this is done, but I cannot get it working because I get a segment fault each time reaches the ‘originate’ command.
Distributor ID: Ubuntu
Description: Ubuntu 16.04 LTS
Here is the blog post I’m referring to:
And here is my dialplan on PasteBin:
GDB backtrace of the core dump on PasteBin: