[asterisk-users] PJSIP does not stop sending invites after call is canceled

[Saint Michael]

Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends  several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP XXXX::50187 --->
CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0
Via: SIP/2.0/UDP xxxxxxx
:50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport
Max-Forwards: 70
To: <sip:xxxxx at xxxxx>
From: "xxxxx"<sip:xxxxx at xxxxx>;tag=a0acbb3e
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
CSeq: 1 CANCEL
User-Agent: Bria 5 release 5.8.3 stamp 102650
Content-Length: 0

PJSIP responds to endpoint

<--- Transmitting SIP response (403 bytes) to UDP:xxxxxx:50187 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
XXXXX:50187;rport=50187;received=XXXX;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "xxxxxx" <sip:xxx at xxxxx>;tag=a0acbb3e
To: <sip:xxxx at xxxxx>;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 CANCEL
Server: Asterisk PBX 13.33.0
Content-Length:  0

the PJISP sends an additional response to endpoint
<--- Transmitting SIP response (419 bytes) to UDP:xxxx:50187 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.16.7.254:50187
;rport=50187;received=xxxxx;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "xxxxx" <sip:xxxx at xxxxx>;tag=a0acbb3e
To: <sip:xxxx at xxxxx>;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 INVITE
Server: Asterisk PBX 13.33.0
Content-Length:  0

to make a long story short, the endpoint sends back an ACK, but after that,
PJSIP keeps sending INVITES to the carrier, which means it did not close
the second leg of the call. If the carrier sends back a 200 OK, there will
be a billing charge, which in case of Mexico is minimum 60 seconds, and the
endpoint will not agree with the charge, resulting in a financial loss for
the Asterisk owner. This is absurd. The second leg must close as soon as a
CANCEL has been received.

The dialplan is only one line
Dial(PJSIP/${EXTEN}@carrier)

Kindly tell me what am interpreting wrong.
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