[asterisk-users] Voice "broken" during calls

Antony Stone Antony.Stone at asterisk.open.source.it
Sat Jun 13 11:20:45 CDT 2020

On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote:

> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
> > Try "sip show peer <peername>" for a phone.

> 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.

> bpi*CLI> sip show peer 0049351xxxxxxx
>   Codecs       : (alaw|ulaw|ilbc|g729|g723|gsm)

That looks a little more standard.



I just got a new mobile phone, and I called it Titanic.  It's already syncing.

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