[asterisk-users] Voice "broken" during calls
Antony.Stone at asterisk.open.source.it
Sat Jun 13 11:16:38 CDT 2020
On Saturday 13 June 2020 at 18:06:23, Michael Keuter wrote:
> So the call used Alaw as Codec.
...which should be excellent quality.
PS: Michael: thanks for the tips regarding "sip show channels" and "sip show
channel <ID>" - I was aware of these for some details, but hadn't realised
they showed the codecs as well. Very useful :)
Your work is both good and original. Unfortunately the parts that are good
aren't original, and the parts that are original aren't good.
- Samuel Johnson
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