UPDATE: I guess it's a good idea to leave this here in case anyone else ever looks for it, but here:
https://code.google.com/archive/p/fusionpbx/issues/409
I found this comment:
"The path is correct because freeswitch automatically looks in 8000,16000,32000, or 48000 depending on the current call."
My problem was that I had only 2 codecs in my allowed codec string and I had my negotiation set to "scrooge" which mean it was forcing the IVR to use a G722 codec even though the leg of the call from which I was calling was on PCMA. When I changed my codec negotiation to generous, the IVR quality improved, and I can now also see when dialling in from externally that it's activating L16@16000 instead of L16@8000 which means that the IVR to external users is also better.
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Hi there,
I am trying to use IVR recordings with different sample rates to see if it will improve the sound. When the IVR reads out WAV files the sound cuts out, and I think perhaps it would be better if I were using the 16000 sample rate files instead of the 8000 ones.
I can't find anywhere to configure this though.
In the logs I can see:
Codec Activated L16@8000hz 1 channels 20ms
I can see that there are log files on the web that have:
Codec Activated L16@16000hz 1 channels 20ms
People have posted their logs to solve *other* issues, so they don't say how they changed the config to that, but I know it must be possible!
https://code.google.com/archive/p/fusionpbx/issues/409
I found this comment:
"The path is correct because freeswitch automatically looks in 8000,16000,32000, or 48000 depending on the current call."
My problem was that I had only 2 codecs in my allowed codec string and I had my negotiation set to "scrooge" which mean it was forcing the IVR to use a G722 codec even though the leg of the call from which I was calling was on PCMA. When I changed my codec negotiation to generous, the IVR quality improved, and I can now also see when dialling in from externally that it's activating L16@16000 instead of L16@8000 which means that the IVR to external users is also better.
------------------------
Hi there,
I am trying to use IVR recordings with different sample rates to see if it will improve the sound. When the IVR reads out WAV files the sound cuts out, and I think perhaps it would be better if I were using the 16000 sample rate files instead of the 8000 ones.
I can't find anywhere to configure this though.
In the logs I can see:
Codec Activated L16@8000hz 1 channels 20ms
I can see that there are log files on the web that have:
Codec Activated L16@16000hz 1 channels 20ms
People have posted their logs to solve *other* issues, so they don't say how they changed the config to that, but I know it must be possible!
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