When I attend-transfer a call, the audio recording of the conversation between the two extensions becomes corrupt after about 1:30 minutes. This is what happens in gsm and wav49 formats. There is no problem in wav format. I noticed that the problem was fixed when I used monitor instead of mixmonitor. but I’m using freepbx and it won’t let me change the extension_additional.conf file.
What do you recommend?
(I also have a sample audio file but I can’t attach it here)
You can make a copy of sub-record-check (or other subroutine that calls MixMonitor), edit it as desired and put it in /etc/asterisk/extensions_override_freepbx.conf .
You could spin up a test system in the cloud, on a VM, or on an old PC to check whether the bug in MixMonitor is still present in current Asterisk versions.
However, why are you recording in a low-quality compressed format? Disk space is cheap. In .wav format, you get about 17,000 hours of recording per terabyte.
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