Why doesn't sipnet understand the hangup signal?

Question

Asterisk

Can anyone suggest what could be the problem.
I'm using Asterisk 1.8.23.1, when calling to a mobile phone, after a conversation (i.e. picking up the phone, talking and hanging up), sipnet does not send a clear signal that the conversation is over. It turns out that the conversation is over on the mobile, but not on the other side. As the other side, an iron sip phone.
Tried on three more sip-providers, the hang-up signal is successfully processed. The settings of other providers are the same.
sipnet trunk settings

[sipnet]
type=friend
insecure=port,invite
disallow=all
allow=ulaw
qualify=yes   
nat=yes ; в значении "no" на sipnete не происходит авторизации
canreinvite=no
fromuser=sip_id
defaultuser=sip_id
context=sipnet_incoming
host=sipnet.ru
secret=password
fromdomain=sipnet.ru
dtmfmode = rfc2833 ; указано в вики sipnet'а, также пробовал inband

Does it make sense to show the entire sip.conf?
I tried sip set debug peer, after you hang up on the mobile, no signal comes, as if the conversation is continuing.
PS I wrote to sipnet's technical support, they answer for a long time, so I'm writing here, maybe just, someone has encountered a problem and knows the solution.
UPD
solution not found, the author left for the New Year holidays, if you have any thoughts, write, I will be glad for any help
, many thanks Soslan Aldatov
, this is what we checked with him, if we are too lazy to read the correspondence (everything that was tried did not help):
tried

exten => _8XXXXXXXXXX,1,Dial(SIP/sipnet/7${EXTEN:1},120,T)
exten => _8XXXXXXXXXX,n,Hangup

Asterisk 1.8.23.1 is located on the router (behind NAT )
the call chain is as follows:
softphone->asterisk->sipnet->mobile picked up the phone hung up_on_mobile->sipnet-(bye does not come from here)>asterisk->softphone
sipnet trunk is registered only in mode nat=yes , nat=route does not help
there is a log of such a call pastebin.com/FKqJE9iz
externip, localnet and externhost are indicated
port 5060 is forwarded (registration passes) there is
sound during the call (the voice goes back and forth)
SIP ALG is not available as a possibility

Answer 1

To get started, just try adding a Hangup:

exten => _8XXXXXXXXXX,1,Dial(SIP/sipnet/7${EXTEN:1},120,T)
exten => _8XXXXXXXXXX,n,Hangup

Answer 2

The problem was this, I fixed the ports for signaling and then bye started coming.
Apparently, when making a call from your aster, some port is reserved under the nat for an outgoing connection. After some timeout, this port closes and bye goes nowhere.