[asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 14.06.2020 um 16:48 schrieb Michael Keuter:

Hi Michael,

> the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß PTMP with 3-10 numbers) is always tied to your DSL account.

I supposed it...

> There is a special "DeutschlandLAN SIP-Trunk Pure" where it does not depend on your DSL account (as it is standard with most other VoIP providers).

OK, I really don't think I want to subscribe this option just to check
if the problem is in my account... :D

Any other suggestion how to find *where* the problem is?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)