[asterisk-users] Voice "broken" during calls
Antony.Stone at asterisk.open.source.it
Sat Jun 13 06:44:07 CDT 2020
On Saturday 13 June 2020 at 13:36:00, Luca Bertoncello wrote:
> Am 13.06.2020 09:30, schrieb Luca Bertoncello:
> Hi again (again)
> I noticed right now another strange detail...
> I made a call using my mobile phone (connected to the Asterisk).
What does that mean? You're making a mobile phone call over the GSM network
to your Asterisk server, or you're using a soft phone application on your
smartphone, which is registered by SIP to your Asterisk server?
Also, where did you make the call *to* ?
> The quality was top...
> Maybe is the problem in a codec used from our phones at homes?
Didn't we already discuss this last year?
> Could someone suggest me how to check the codec used by my mobile phone
> and the codec used by the phones at home?
Look at the verbose log file and search for "transcoding".
Also, do a SIP packet trace at the start of the call and see which codecs are
announced by each side and then what gets agreed on (I don't think this gets
logged by Asterisk, so you need to look at the SIP negotiation itself).
I'm not impossible, just highly implausible.
Please reply to the list;
please *don't* CC me.
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