Why does freeswitch have sound problems?

Question

FreeSWITCH

Hello. There is such a scheme:
SIP-client <-> freeswitch <-> asterisk <-> chan_mobile <-> cell phone via bluetooth.
Asterisk has a built-in extension 105 and a trunk to freeswitch. All calls to the asterisk are redirected to the cell phone. Incoming cell phones go to freeswitch.
The problem looks like this: when making an outgoing call from a SIP client, there is no incoming sound. There is a crack or nothing at all. There is outgoing sound (the sound comes to the cell phone). At the same time, if you connect a softphone instead of asterisk, then the sound quality is at its best. In all directions. If you call directly from the asterisk (having connected with 105 extension), then everything is fine with the sound too.
I took a dump, on the asterisk the sound when calling through the entire chain is still normal. Broken already comes to freeswitch.
If you call between 105 and SIP-client, then everything is fine with the sound too.
I can't figure out at what stage it hits the fan :( aster config pastebin.com/EYt3SGEv
freeswitch dialplan piece pastebin.com/kaN1z7UP It's mostly standard (default). There is also copy-paste in the freeswitch dialplan: pastebin.com/ B3kFSWTP I haven't figured out how to make calls to non-digital extensions otherwise vgc1-2 are clients, vgg and gate are gateways.

Answer 1

stupid copy-paste usually leads to this and
can you explain why in the bridge "absolute_codec_string='g722'"?