So I tried all of the combinations (restarted after modifying of course); results:
sip_as_rtmp_use_new_engine=false
****
streamnotfound when I tried to play the stream
force_rtmp_audio_codec=speex16
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no sound (therefore no change when compared to the original settings)...
Hmm, for some reason if I use the rtmp://dev.***.com as you did, it works, but with the other RTMP server (rtmp://staging.***.com), I'm not receiving the sound as I described earlier.
I need this to work with staging
Ok so I made some adjustments, the RTMP looks like it's capturing it correctly, but I cannot hear any sound, although the call is exactly 1 minute 16 seconds which it should be. Please try - I set up a dummy call that repeats "One two three" for 76 seconds.
Ok I fixed the RTMP and it works now. Only thing is that getStatus now returns the stream ID, but on the other server it returns "ESTABLISHED" - is that because I have a newer version on this server?
If it's the problem just with RTMP server not receiving the stream, why does the getStatus say "NO SUCH CALL" after 2 seconds of the call failing although the SIP call should take 100 seconds?
Hi,
so we tried to duplicate what we had working on our dev server to our live, but the call doesn't seem to be established correctly (after the first getStatus, it already returns "NO SUCH CALL" although the call should take over 100 seconds). I am sending an e-mail to logs@ now as well...
Ok so I imported the keys successfully, the host:8888 now shows green https instead of crossed out red https which it was showing before, but a REST call to https://host:8888/RESTCall/call now returns 500 with empty body (no error message)