[asterisk-users] Voice "broken" during calls
lucabert at lucabert.de
Wed Jun 17 08:07:03 CDT 2020
Am 17.06.2020 14:37, schrieb Karsten Wemheuer:
> The product is "All-IP" and not the SIP trunk, right?
> The call starts normally and after about 15 minutes the quality is
No, current we have Magenta Zuhause. Tomorrow we'll change to
DeutschlandLAN IP (business contract).
The quality is disturbed from the first second...
I had the problem, that the connection will be *dropped* after 15
minutes, and I solved it with "session-timers = refuse"
(lucabert at lucabert.de)
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