Trubble with 183 Session progress

OS: Debian 7.7
Asterisk: 13.1.0
FreePBX: 12.0.22
Building from source
After starting the FreePBX, I prescribe several subscribers with the system. Calls between subscribers are fine. I use chan_sip. As soon as I dialed the number nonexistent subscriber
I always was heard information:
"silence / 1 & can not-complete-as-dialed & check-number-dial-again, noanswer"
Asterisk (asterisk -rvvvvvv) writes about it, but I hear only at RBT then quite normal error 503.
Next I tried to replace Ttr to Ttm, the system writes that there is music from the “default”, but I hear only the RBT.
Began testing using tshark, and saw me back 180 RING instead of 183 session progress + RTP.
Then I switched subscriber chan_pjsip - a miracle, everything fell into place, but my religion does not allow me to use chan_pjsip, as pjsip may be underdeveloped.
When I tried to introduce chan_pjsip we pulled other problems, so at the time had yet to leave it.

debug:

[quote][code]<— SIP read from UDP:172.16.11.103:5063 —>
INVITE sip:1706@172.16.103.196 SIP/2.0
Via: SIP/2.0/UDP 172.16.11.103:5063;branch=z9hG4bK1559491095
From: “1704” sip:1704@172.16.103.196;tag=1116255126
To: sip:1706@172.16.103.196
Call-ID: 2418212244@172.16.11.103
CSeq: 2 INVITE
Contact: sip:1704@172.16.11.103:5063
Authorization: Digest username=“1704”, realm=“asterisk”, nonce=“2ff01588”, uri="sip:1706@172.16.103.196", response=“d60aaaba510e29d26652cf6d6bb2dd5f”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T22P 7.72.14.6
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 296

v=0
o=- 20192 20192 IN IP4 172.16.11.103
s=SDP data
c=IN IP4 172.16.11.103
t=0 0
m=audio 11790 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to 172.16.11.103:5063 (no NAT)
Using INVITE request as basis request - 2418212244@172.16.11.103
Found peer ‘1704’ for ‘1704’ from 172.16.11.103:5063
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.11.103:11790
Looking for 1706 in from-internal (domain 172.16.103.196)
sip_route_dump: route/path hop: sip:1704@172.16.11.103:5063

<— Transmitting (no NAT) to 172.16.11.103:5063 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.11.103:5063;branch=z9hG4bK1559491095;received=172.16.11.103
From: “1704” sip:1704@172.16.103.196;tag=1116255126
To: sip:1706@172.16.103.196
Call-ID: 2418212244@172.16.11.103
CSeq: 2 INVITE
Server: FPBX-12.0.21(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1706@172.16.103.196:5060
Content-Length: 0

<------------>
– Executing [1706@from-internal:1] ResetCDR(“SIP/1704-000002f7”, “”) in new stack
– Executing [1706@from-internal:2] NoCDR(“SIP/1704-000002f7”, “”) in new stack
– Executing [1706@from-internal:3] Progress(“SIP/1704-000002f7”, “”) in new stack

<— Transmitting (no NAT) to 172.16.11.103:5063 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.11.103:5063;branch=z9hG4bK1559491095;received=172.16.11.103
From: “1704” sip:1704@172.16.103.196;tag=1116255126
To: sip:1706@172.16.103.196;tag=as1ded75ed
Call-ID: 2418212244@172.16.11.103
CSeq: 2 INVITE
Server: FPBX-12.0.21(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1706@172.16.103.196:5060
Content-Length: 0

<------------>
– Executing [1706@from-internal:4] Wait(“SIP/1704-000002f7”, “1”) in new stack
– Executing [1706@from-internal:5] Progress(“SIP/1704-000002f7”, “”) in new stack
– Executing [1706@from-internal:6] Playback(“SIP/1704-000002f7”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/1704-000002f7> Playing ‘silence/1.slin’ (language ‘ru’)
– <SIP/1704-000002f7> Playing ‘cannot-complete-as-dialed.slin’ (language ‘ru’)

<— SIP read from UDP:172.16.102.101:5062 —>

<------------->
– <SIP/1704-000002f7> Playing ‘check-number-dial-again.slin’ (language ‘ru’)
– Executing [1706@from-internal:7] Wait(“SIP/1704-000002f7”, “1”) in new stack
– Executing [1706@from-internal:8] Congestion(“SIP/1704-000002f7”, “20”) in new stack

<— Reliably Transmitting (no NAT) to 172.16.11.103:5063 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.11.103:5063;branch=z9hG4bK1559491095;received=172.16.11.103
From: “1704” sip:1704@172.16.103.196;tag=1116255126
To: sip:1706@172.16.103.196;tag=as1ded75ed
Call-ID: 2418212244@172.16.11.103
CSeq: 2 INVITE
Server: FPBX-12.0.21(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (from-internal, 1706, 8) exited non-zero on ‘SIP/1704-000002f7’
– Executing [h@from-internal:1] Hangup(“SIP/1704-000002f7”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1704-000002f7’

<— SIP read from UDP:172.16.11.103:5063 —>
ACK sip:1706@172.16.103.196 SIP/2.0
Via: SIP/2.0/UDP 172.16.11.103:5063;branch=z9hG4bK1559491095
From: “1704” sip:1704@172.16.103.196;tag=1116255126
To: sip:1706@172.16.103.196;tag=as1ded75ed
Call-ID: 2418212244@172.16.11.103
CSeq: 2 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘2418212244@172.16.11.103’ Method: ACK

<— SIP read from UDP:172.16.11.103:5063 —>
[/code][/quote]

Myself asked, himself will answer:

Well, in general, as found necessary to include in the settings FreePBX, Other SIP Settins write: progressinband = yes
I think that this is a bug FreePBX, Asterisk does not apply to.