Freepbx voice lag after transfer


We are having the voice lab issue after call transfer:

A person “A” calls our external partner “B”: partner B then automatically redirects call to our company “X”, then X hears what the person “A” wants and retransfer the call to a specific phone number at the office of the partner B.

So this chain is rather long

Person A-> Partner B-> our company X-> Partner B

Only at the last step the delay starts.

The lines remain allocated on FreePBX (our company) until the end of the call between the Person A and the number to which we have made the transfer.

Can you suggest where to start troubleshooting and what to look for?

Thanks in advance!

Do you need to stay in the media path (recording, whispering, features codes, etc.)?

What technology is used for each hop?

What is the difference between redirect and transfer, and how are these implemented (e.g timed break recall on analogue, SIP 302, SIP REFER, or SIP INVITE then REFER/Replaces)?

By how much does the voice lag?

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