Application Record: How to manipulate the volume of the wav-file?


I’m using the application Record to save a sound file. For example, the application Record has the following passed parameters: hello.wav,180,qk.

The application works well. However, the saved sound file has a low volume!

It is possible to manipulate the volume of the recorded sound file?
What technology does the application Record used?
Does asterisk have an alternative for the application Record?

I’m using the asterisk version 13.1.


Best Regards

MixMonitor has the following options

v - Adjust the heard volume by a factor of x (range -4 to 4)
V - Adjust the spoken volume by a factor of x (range -4 to 4)
W - Adjust both, heard and spoken volumes by a factor of x (range -4 to 4)

Also you can use the sox tool to increase the volume

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Record will record the sound at exactly the level at which it was digitised at the boundary of the digital part of the network. It won’t adjust the volume at all.

For fully VoIP systems, the boundary is the phone. For ISDN, it is likely to be the line card in the remote party’s local exchange. For analogue, it will be the FXO or FXS module in your system. It should play back, through Asterisk, at the same volume as a direct call from the same source.

Thank you for the information.

I have found the problem.

The recorded SIP-call is using the g.722 codec and the application Record save the sound file in wav format! In this case the sound file has a lower volume.

If the recorded SIP call is using the g.711 codec everything works fine!

I want to try with other sound file formats or has anybody an idea to solve the problem.

Which sound file format supports asterisk?

All Asterisk will do is run it through its g722 codec. If that has scaling error, it will be reflected both in the recording and in any G.711 only phones directly bridged through that codec. I would find it surprising that such a problem had never been reported before, but it is possible.

Are you sure that the G.722 source is scaling the audio correctly when using G.722?

There is a conversion tool in Asterisk. Try converting ulaw or alaw to g722 and back. If the result is the correct volume, without clipping, it would confirm that the problem is in your incoming G.722.

Thank you for your immediately answer! I will check it with the file converter.

Let me explain in short words what I’m doing.
First I record a a sound file with the application record. Later on, I play back this sound file with the application Playback to a SIP-target (loudspeaker) with g.722 codec.

If i recorded with the g.722 codec the playback has an lower volume.
If I recorded with the g.711 codec the playback has the same volume.
The refernce volume is a SIP connection between both SIP entiy.

In the meantime, I have tested with other sound file formats (g722 and gsm) but I get the same results.