My outbound calls are recorded, but they are very slow, so the sound is pitched down. When I download them an increase the speed in VLC, they play ok. This seems to only happen with outbound calls. I’m also using opus, if that might be contributing to the problem. I’m on asterisk 16.0.0.
I found this issue that was closed, but it seems like it is referencing inbound calls, which seem to be OK for me.
What steps should I go through to solve this problem?
Ok, so I have confirmed it is related to calls using the opus codec. Calls record without a problem when outbound calling using a different codec. Is this indeed an asterisk issue? Should I move this topic over there?
In the larger sense, it sounds like the same problem you’d have with a stereo recording playing it through a mono processor. Each channel gets played in succession, so it sounds like “half speed”.
I must admit I don’t have a lot of experience with audio. But what you’re saying makes a little bit of sense. I guess I just don’t know what would cause it to do that.
Also, I wanted to update asterisk to see if they had already addressed this issue, but I’m having difficulty finding documentation on how to do this. I’m currently on 16.0.0, but there is a newer version available (16.2.1). I’ve already tried using asterisk-version-switch, but that always brings me to 16.0.0. I’ve updated everything on the server, to no avail. Is it possible that 16.2.1 just isn’t supported on FreePBX yet?
It’s writing this into slin48 how are you playing this back? You said VLC. The issue with slin is that most programs default to thinking this is slin8 and thus play then at a sample rate of 8khz instead of 48khz. Which would sound slowed down. You need a program that can accurately understand that the slin file is 48khz not 8khz.