Multifon and Asterisk. How to solve the problem of intermittent voice?

Question

Asterisk

There is an Asterisk server. It has a multiphone connected. The problem is that with an incoming call, the interlocutor is intermittently heard, but I can be heard without problems. As a Mikrotik gateway. The udp and tcp ports are forwarded, sip alg disabled (does not help), the range of rtp ports is 5000-20000.
Googling included jitterbuffer=yes and trunktimestamps=yes in iax.conf ulaw
and alaw codecs. Internet channel 50 megabits, not loaded.
Multiphone peer settings:

[my_multifon_user](!)
type=peer
videosupport=no
host=193.201.229.35      
insecure=invite,port     
fromdomain=megafon.ru
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
qualify=yes
transport=udp
nat=force_rport,comedia
dtmfmode=inband
relaxdtmf=yes
port=5060
context=_sip

The most interesting thing is that when you call yourself through a multiphone, you can hear it chic.
What else to look at and how to solve the problem?

Answer 1

The multiphone itself also periodically falls off. sip show registry displays Request sent and time out

Answer 2

In sip.conf specify:

externip=ваш_внешний_ip
localnet=ваша_локальная_сеть/префикс локальной сети

During a call, see
Is there any packet loss? How much %?
Why did you include something in iax.conf if there is no iax? Better vice versa, disable all modules you don't use and delete their default configs.

Answer 3

It is worth setting up a shaper for traffic in Mikrotik itself and disabling unnecessary things in Mikrotik itself.