The call does not come for a long time and flies out, from what?

Question

Telephony

Hey!

Please help, I've been struggling for a week now, I can't solve the problem.

Yes:
Centos 7
Asterisk 18.11.1

When I call an internal number, according to the log, I see that the request goes immediately, instantly, and then some long pause of 15-20 seconds, sometimes more, and only then a call. Call from internal number to internal. I read that sometimes there is a problem due to DNS. Changed everything to IP. Even the caching DNS server installed its own, but nothing has changed.

cat pjsip.conf

[global]
user_agent=Phone
allow_reload=yes
nat=force_rport,comedia

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:6677
external_media_address=26.89.31.175
external_signaling_address=26.89.31.175

[endpoint_tpl](!)
type=endpoint
transport=transport-udp
context=sip_local
dtmf_mode=rfc4733
media_encryption=no
aggregate_mwi=yes
use_avpf=no
message_context=messages
disable_direct_media_on_nat=yes
rtcp_mux=no
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
direct_media=no
language=ru
disallow=all
allow=alaw
allow=gsm
media_address=26.89.31.175

[auth_tpl](!)
type=auth
auth_type=userpass

[aor_tpl](!)
type=aor

Log before call

И после этого ожидание и пауза... Помогите плиз...

<--- Received SIP request (746 bytes) from UDP:100.65.148.136:44982 --->
INVITE sip:[email protected]:6677;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:44982;branch=z9hG4bK-524287-1---f0467f055bf50947;rport
Max-Forwards: 70
Contact: <sip:[email protected]:29959;transport=UDP>
To: <sip:[email protected]:6677>
From: <sip:[email protected]:6677;transport=UDP>;tag=a6449c6d
Call-ID: QGte_MV5DwfTRG38U_pS9w..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.17.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 179

v=0
o=Zoiper 236247790 1 IN IP4 93.185.36.44
s=Z
c=IN IP4 93.185.36.44
t=0 0
m=audio 29737 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (502 bytes) to UDP:100.65.148.136:44982 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.12:44982;rport=44982;received=100.65.148.136;branch=z9hG4bK-524287-1---f0467f055bf50947
Call-ID: QGte_MV5DwfTRG38U_pS9w..
From: <sip:[email protected]>;tag=a6449c6d
To: <sip:[email protected]>;tag=z9hG4bK-524287-1---f0467f055bf50947
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1649863082/99d468c62db4792a526e75957351f923",opaque="32530c5a4cef852a",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.11.1
Content-Length:  0


<--- Received SIP request (356 bytes) from UDP:100.65.148.136:44982 --->
ACK sip:[email protected]:6677;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:44982;branch=z9hG4bK-524287-1---f0467f055bf50947;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=z9hG4bK-524287-1---f0467f055bf50947
From: <sip:[email protected]:6677;transport=UDP>;tag=a6449c6d
Call-ID: QGte_MV5DwfTRG38U_pS9w..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1046 bytes) from UDP:100.65.148.136:44982 --->
INVITE sip:[email protected]:6677;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:44982;branch=z9hG4bK-524287-1---12deb004a21b4555;rport
Max-Forwards: 70
Contact: <sip:[email protected]:29959;transport=UDP>
To: <sip:[email protected]:6677>
From: <sip:[email protected]:6677;transport=UDP>;tag=a6449c6d
Call-ID: QGte_MV5DwfTRG38U_pS9w..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.17.3-mod
Authorization: Digest username="202",realm="asterisk",nonce="1649863082/99d468c62db4792a526e75957351f923",uri="sip:[email protected]:6677;transport=UDP",response="32bef6ed9342518be95b2f6422079123",cnonce="0c6609e835f61208f172c8b02ea63f43",nc=00000001,qop=auth,algorithm=md5,opaque="32530c5a4cef852a"
Allow-Events: presence, kpml, talk
Content-Length: 179

v=0
o=Zoiper 236247790 1 IN IP4 93.185.36.44
s=Z
c=IN IP4 93.185.36.44
t=0 0
m=audio 29737 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (310 bytes) to UDP:100.65.148.136:44982 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.12:44982;rport=44982;received=100.65.148.136;branch=z9hG4bK-524287-1---12deb004a21b4555
Call-ID: QGte_MV5DwfTRG38U_pS9w..
From: <sip:[email protected]>;tag=a6449c6d
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: Asterisk PBX 18.11.1
Content-Length:  0

I don't know where to look, what to do. I reinstalled asterisk several times.