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@brar
asterisk

Почему транк прова игнорит Hangup?

Номер из серии 8800ххххххх.
Сократил экстен до минимума. 
exten => 8800ххххххх,1,Set(CALLERID(name)=8800)
    same => n,Dial(SIP/3899,5,tT) 
    same => n,Hangup()


После 5 секунд Hangup в логе выполняется, но сразу идет новый дозвон Dial(SIP/3899,5,tT) . Для вызывающего абонента провайдерский MOH играет бесшовно. Создается новый канал. То есть бесконечный цикл дозвона, до тех пор пока не возьмут трубку.
Такой проблемы с номерами 7495 нет, несмотря на то, что экстены идентичны.
На всякий случай прикладываю дебаг, в котором:
PROVIDER-IP - айпи-адрес сервера провайдера.
192.168.55.8 - астер.
192.168.77.7 - тлф.
9161234567 - номер звонящего абонента на 8800.
Дебаг от начала Hangup и до нового дозвона:

Nobody picked up in 5000 ms 
Scheduling destruction of SIP dialog '58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.77.7:51979:
CANCEL sip:3899@192.168.77.7:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
Max-Forwards: 70
From: "BOUNDs8800" <sip:9161234567@192.168.55.8>;tag=as7a646dd0
To: <sip:3899@192.168.77.7:51979;transport=tcp>
Call-ID: 58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060
CSeq: 102 CANCEL
User-Agent: BOUNDsN9NE
Content-Length: 0
---
Scheduling destruction of SIP dialog '58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060' in 32000 ms (Method: INVITE)
    -- Executing [8800xxxxxxx@in-BOUNDs: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:9161234567@192.168.55.8>;tag=as7a646dd0
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Call-ID: 58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060
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:9161234567@192.168.55.8>;tag=as7a646dd0
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Call-ID: 58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060
Date: Mon, 14 Oct 2019 19:44:49 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:3899@192.168.77.7:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:3899@192.168.55.8>;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:3899@192.168.77.7:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK4ea908d8
Max-Forwards: 70
From: "BOUNDs8800" <sip:9161234567@192.168.55.8>;tag=as7a646dd0
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200076f8d92a9-fd955446
Contact: <sip:9161234567@192.168.55.8:5060;transport=tcp>
Call-ID: 58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060
CSeq: 102 ACK
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '58145cfc56bddc313d44cfb514dee293@192.168.55.8:5060' in 32000 ms (Method: INVITE)
  == Using SIP RTP CoS mark 5
       > 0x7f9c74022b30 -- Strict RTP learning after remote address set to: PROVIDER-IP:15918
    -- Executing [8800xxxxxxx@in-BOUNDs:1] Set("SIP/BOUNDs_01010-000000bb", "CALLERID(name)=BOUNDs8800") in new stack
    -- Executing [8800xxxxxxx@in-BOUNDs: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:3899@192.168.77.7:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>
Contact: <sip:9161234567@192.168.55.8:5060;transport=tcp>
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
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:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
Date: Mon, 14 Oct 2019 19:44:53 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:3899@192.168.77.7:51979;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:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
Date: Mon, 14 Oct 2019 19:44:53 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:3899@192.168.77.7:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:3899@192.168.55.8>;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:3899@192.168.77.7:51979;transport=tcp>
    -- SIP/3899-000000bc is ringing
Scheduling destruction of SIP dialog '7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.77.7:51979:
CANCEL sip:3899@192.168.77.7:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
CSeq: 102 CANCEL
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060' 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:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
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:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
Date: Mon, 14 Oct 2019 19:44:55 GMT
CSeq: 102 INVITE
Server: Cisco-CP7906G/8.3.0
Contact: <sip:3899@192.168.77.7:51979;transport=tcp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "IT" <sip:3899@192.168.55.8>;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:3899@192.168.77.7:51979;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.55.8:5060;branch=z9hG4bK50a4603e
Max-Forwards: 70
From: "BOUNDs8800" <sip:9161234567@192.168.55.8>;tag=as4df39872
To: <sip:3899@192.168.77.7:51979;transport=tcp>;tag=002155d59a5200092062c48a-caebb191
Contact: <sip:9161234567@192.168.55.8:5060;transport=tcp>
Call-ID: 7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060
CSeq: 102 ACK
User-Agent: BOUNDsN9NE
Content-Length: 0


---
Scheduling destruction of SIP dialog '7e5a927271b54eee03a2522b593cc1e8@192.168.55.8:5060' in 32000 ms (Method: INVITE)

Вопрос: как заставить астериск НЕ совершать новый дозвон по истечению таймаута в Dial?
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