I'm currently working with LumiSoft's SIP stack, and I'm able to successfully register an extension on my FreePBX box and make a call to another softphone. All I need to do now is stream a WAV file (or RAW, or whatever will work) through the call so the recipient can hear it.
I've seen the RTP Audio tutorial he has on his site -http://www.lumisoft.ee/lsWWW/download/downloads/Examples/
It looks like you can put in 2 end points, (the local box, and the PBX host), and "stream" the file, but how can I inject it into the call between 2 extensions? I'm very new to VOIP, SIP, and all of that so I feel like I'm missing something simple.
Has anyone here done something like this before? I'm really looking for a push in the right direction.
Turns out, the easiest way to solve this was to install VoiceMeeter (http://vb-audio.pagesperso-orange.fr/Voicemeeter/index.htm) and then just play audio through the speaker as normal. Once I register my SIP and make an outgoing call, once someone picks up I can play my WAV straight through the speakers and the callee hears it with no issue!