How to determine from which side the channel broke?

Question

Asterisk

The architecture of the system in general terms looks like this:

Softswitch on FreePBX, on which SIP trunks from telecom providers are built; behind it is an asterisk, on which the tubes of subscribers (different virtual exchanges) are registered. Those. the softswitch acts as a transit node.

The client complains that conversations break through. Other tennants on the same asterisk work fine.

The end of the conversation in the log looks like this:

pbx.c: Spawn extension (ext-trunk, tdial, 10) exited non-zero on 'SIP/3101198-00010d46'

I can't find the ext-trunk context anywhere. Can you please tell me how to determine which side is the gap? From the side of the asterisk, where all the clients live, or from the side of the telecom provider?

It is very difficult to debug a trunk due to the large number of calls passing through it.

Answer 1

And yet you have to debug.
It is not necessary to enable debugging in the asterisk. Use tcpdump, or better yet, sngrep.
Save a dump over a period, say 2-3 hours, to a file. Then open with sngrep, filter by the desired number and you will see who sent the BYE.

Answer 2

Without monitoring, one can only guess.
Try to deploy SIP3.