Asterisk sip debug

Hello everyone…
I have problems with an Asterisk pbx. The pbx has 2 ISDN lines on a Digium card. The quality of the voice during calls varies greatly from one call to another and often it gets to the point that you cannot understand what the other person is saying. For a few months now, at times, there has even been a complete drop in the call in progress. I wanted to know if by providing data such as configuration files, logs and PCAP dump files someone could provide me with paid help.
I have already tried changing the ISDN card and the switchboard completely.
Thank you

When you say switchboard, do you mean you also changed all the wires in case any were cut? What about checking earth/ground on your equipment?

yesterday with an electrician we ran some new cables to separate the telephone network from the PC network. Although I couldn’t put a separate cable for each phone and therefore I will have to add 3 or 4 more switches…
Tomorrow I’m going to do the tests…
In your opinion, is it appropriate to separate the networks with a router to have different classes (I don’t have a switch with VLAN management) or can having two separate switches already have positive effects?

Separate “smart” switches in the telco closet are great but putting little “dumb” ones out and about in the office where users can plug things in, loop them up, etc. … not so great. :frowning:

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