Why is Provo trunk ignoring Hangup?

Question

Asterisk

Series number 8800хххххх.
Reduced the exten to a minimum.

exten => 8800ххххххх,1,Set(CALLERID(name)=8800)
    same => n,Dial(SIP/3899,5,tT) 
    same => n,Hangup()

After 5 seconds, Hangup in the log is executed, but a new dial-up is immediately Dial(SIP/3899,5,tT) . For the caller, the provider's MOH plays seamlessly. A new channel is being created. That is, an endless dialing cycle until they pick up the phone.
There is no such problem with 7495 numbers, despite the fact that the extents are identical.
Just in case, I attach a debug, in which:
PROVIDER-IP - IP address of the provider's server.
192.168.55.8 - aster.
192.168.77.7 - tel.
9161234567 - the number of the caller on 8800.

Debug from Hangup start to new call:

Nobody picked up in 5000 ms 
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.77.7:51979:
CANCEL sip:[email protected]:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
Max-Forwards: 70
From: "BOUNDs8800" <sip:[email protected]>;tag=as7a646dd0
To: <sip:[email protected]:51979;transport=tcp>
Call-ID: [email protected]:5060
CSeq: 102 CANCEL
User-Agent: BOUNDsN9NE
Content-Length: 0
---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
    -- Executing [[email protected]:3] Hangup("SIP/BOUNDs_01010-000000b9", "") in new stack
  == Spawn extension (in-BOUNDs, 8800xxxxxxx, 3) exited non-zero on 'SIP/BOUNDs_01010-000000b9'

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
From: "BOUNDs8800" <sip:[email protected]>;tag=as7a646dd0
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:49 GMT
CSeq: 102 CANCEL
Server: Cisco-CP7906G/8.3.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
From: "BOUNDs8800" <sip:[email protected]>;tag=as7a646dd0
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:49 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:[email protected]:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 192.168.77.7:51979:
ACK sip:[email protected]:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
Max-Forwards: 70
From: "BOUNDs8800" <sip:[email protected]>;tag=as7a646dd0
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Contact: <sip:[email protected]:5060;transport=tcp>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
  == Using SIP RTP CoS mark 5
       > 0x7f9c74022b30 -- Strict RTP learning after remote address set to: PROVIDER-IP:15918
    -- Executing [[email protected]:1] Set("SIP/BOUNDs_01010-000000bb", "CALLERID(name)=BOUNDs8800") in new stack
    -- Executing [[email protected]:2] Dial("SIP/BOUNDs_01010-000000bb", "SIP/3899,5,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16024
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.77.7:51979:
INVITE sip:[email protected]:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>
Contact: <sip:[email protected]:5060;transport=tcp>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: BOUNDsN9NE
Date: Mon, 14 Oct 2019 19:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322

v=0
o=root 2098500054 2098500054 IN IP4 192.168.55.8
s=Asterisk PBX 15.7.1
c=IN IP4 192.168.55.8
t=0 0
m=audio 16024 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/3899

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:53 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:[email protected]:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:53 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:[email protected]:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected]:51979;transport=tcp>
    -- SIP/3899-000000bc is ringing
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.77.7:51979:
CANCEL sip:[email protected]:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>
Call-ID: [email protected]:5060
CSeq: 102 CANCEL
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (in-BOUNDs, 8800xxxxxxx, 2) exited non-zero on 'SIP/BOUNDs_01010-000000bb'

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:55 GMT
CSeq: 102 CANCEL
Server: Cisco-CP7906G/8.3.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from TCP:192.168.77.7:51979 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: [email protected]:5060
Date: Mon, 14 Oct 2019 19:44:55 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:[email protected]:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 192.168.77.7:51979:
ACK sip:[email protected]:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:[email protected]>;tag=as4df39872
To: <sip:[email protected]:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Contact: <sip:[email protected]:5060;transport=tcp>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)

Question: how to force the asterisk NOT to make a new call after the timeout in Dial?

Answer 1

Try 19 or 21 codes.
https://wiki.asterisk.org/wiki/display/AST/Hangup+...
exten => 8800хххххх,1,Set(CALLERID(name)=8800)
same => n,Dial(SIP/3899,5,tT)
same => n,Hangup(19)