Different sample rate for outbound call recording

The codec that is mostly supported (and negotiated) for my calls is G711 which has 8khz sample rate. So both channels for all calls are getting recorded in 8khz. Is it possible to ask Asterisk to record outbound channel in 16khz (i.e. in G711.1) with keeping original frequency information present (and not just resampling 8khz audio to 16khz)?

Unfortunately, I don’t have experience with Asterisk directly, another team is operating Asterisk servers. I’m working on a speech recognition solution so getting wideband audio is important for recognition quality, even getting one channel in 16khz would be great.


This is not supported, all paths would result in upsampling.

Thanks for the answer :slight_smile: What’s the appropriate place to file a feature request? JIRA https://issues.asterisk.org/jira/secure/Dashboard.jspa?

We don’t currently accept feature requests without patches. If this is something you need you can try using the bounty program[1] to elicit interest, but I’m not sure from an implementation perspective what all would be required. Things aren’t really architected for this…

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

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