What is a “fracture”? Should “exits” be “there exist”? What does “fels” mean?
Also, there is likely to be far too little informatiion to work out what is wrong, even when those points are clarified.
What channel technology are you using (e.g. DAHDI analogue, H.323 etc.)? If there is more than one implentation of the technology, which one? Please provide relevant configurations and logs.
extensions.conf and either sip.conf or pjsip.conf, depending on which SIP driver you are using.
Note that, as you use the term “sip trunk”, which is not an Asterisk term, you may be using FreePBX, in which the relevant parts of these files is likely to included from the main file.
Note that interruption may just indicate that something is overloaded.
I use Asterisk 16.5. Sip.conf part i attached.
extensions.conf part is simple, just adding incoming call into queue.
There is no overload on RAM, Network, CPU, etc.
Note: In interrupted calls lost send packets. I see it from these command ‘sip show cahnnelstats’ and also listening recorded file.
Does the sound stop at a particular time into a particular part of the call. Does it restart on its own? If so has something changed in the call before it restarts.
Basically, though, the biggest part of answering this question is going to be working out what is going wrong, so we will need as much information as possible.
Sometimes sound stop at particular time (1, 2 second) then continues sound. It is occure several times on one call.
And sometimes sound does not restarted, sound stoppped.
…that subnet looks like your problem, if you are able to correlate the recording outages to the increase in Lost packet % for that call. It could be you have a bad switch, overloaded router, or something else specific to that portion of your network.