[asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 13.06.2020 um 18:20 schrieb Antony Stone:

Hi

>> bpi*CLI> sip show peer 0049177xxxxxxx
>>   Codecs       :
>> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|
>> slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|t
>> estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk
>> |silk|silk)
> 
> That strikes me as somewhat unlikely.

Too much things, isn't it?

>> bpi*CLI> sip show peer 0049351xxxxxxx
>>   Codecs       : (alaw|ulaw|ilbc|g729|g723|gsm)
> 
> That looks a little more standard.

The questions are:

1) why the mobile phone, with "too many things" has a better quality
2) where can I change these settings?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)