[asterisk-bugs] [JIRA] (ASTERISK-20780) Asterisk responds to SIP CANCEL with 481 Call/Transaction Does Not Exist

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-20780?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-20780:
------------------------------------

    Assignee: mitja  (was: Rusty Newton)
      Status: Waiting for Feedback  (was: Triage)

I looked over the RFC 3261 and saw that
"The following procedures are used to construct a CANCEL request.  The
   Request-URI, Call-ID, To, the numeric part of CSeq, and From header
   fields in the CANCEL request MUST be identical to those in the
   request being cancelled, including tags. ..."

All of those headers appear identical between the INVITE and associated CANCEL in your capture and debug, except the "To:" header on the CANCEL has a "tag=as790acac3", whereas the INVITE's "To:" does not.

In this case it looks like Asterisk is responding correctly to your provider sending a malformed CANCEL request.

I'd recommend contacting your provider and seeing if they can fix the issue with their INVITE and CANCEL's not matching up.


                
> Asterisk responds to SIP CANCEL with 481 Call/Transaction Does Not Exist
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-20780
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20780
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.15.0
>            Reporter: mitja
>            Assignee: mitja
>         Attachments: extensions.conf, scenario1.pcapng, Scenario1.txt, Scenario2.txt, senario1
>
>
> Hi. 
> First scenario:
> I was testing sip trunk on asterisk 1.8, when incoming call is initiated and outside user then hangup without answering, phone will not stop ringing. The provider send me a Cancel request (Request-Line: CANCEL sip:77XXXXXX at 192.168.0.11:5060 SIP/2.0) but then asterisk give me " Status-Line: SIP/2.0 481 Call/Transaction Does Not Exist" error. In these case we have to restart asterisk to free a channel. In earlier versions of asterisk  (1.4,1.6) with same settins these problem never occurred, but only in asterisk 1.8.
> Second scenario:
> When incoming call is initiated and enduser answers the call, than all other calls hangup normaly, until asterisk is reloaded or sip trunk is reregistered.
> I searched the internet for a few days now but I did not found nothing, I did try some patches for reinvite issues but nothing worked.
> Thanks for your answer
> bye, Mitja

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