Soundfiles guidance

I found soundfiles for asterisk in “sln16” format.
What do I do with them if my preferred codecs are: Opus, G.722, G.711.

Do I need to convert this files in the format of every codec I want to use?

If yes, is there an “easy” way to do that?

Of what I read I need to convert those files into .wav and from there into the preferred format?

Thanks in advance,


Depending on your cpu overhead this may or may not be necessary. If you need to transact in other codecs Asterisk will likely transcode and increase the cpu usage. If you move a recording into your PBX using the system recording module, you can transcode into the different codecs that FreePBX offers at the time of import.

Although asterisk is good at converting to and from slin , you can if you wish, download from

and extract to your correct language directory, many formats, quite a few languages, you need the ‘core’ sounds and the ‘extra sounds’ for completeness

Thank you for all the answers. Unfortunately my language is not provided there but I found it in the named format.
So if i understand correctly I can convert the sln16 files by uploading it via webinterface (Sound languages module) and then selecting the codecs I want it to be converted into like shown in the wiki:

Currently supported formats are: alaw, g722, gsm, sln, sln16, sln48, ulaw, wav.

If I´m not mistaken, no transcoding needs to be done by asterisk in that case.

No run time transcoding of the sound files will be needed if you pre-transcoded them into all those formats. Run time transcoding may be needed for other reasons, like recording calls, whispering, conferences, in band tone generation, silence detection, in band DTMF decoding, etc.

You will need slin16, not slin, as the starting point, to achieve the theoretically best results from G.722, but you did say slin16 in the original posting.

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