Saint Michael
2020-May-16 14:42 UTC
[asterisk-users] PJSIP does not stop sending invites after call is canceled
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200516/9d2c67ad/attachment.html>