[asterisk-bugs] [JIRA] (ASTERISK-26616) T38 calls not being hung up correctly

[Morten Tryfoss (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26616?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Morten Tryfoss updated ASTERISK-26616:
--------------------------------------

    Attachment: full_debug_log.txt
                calls.pcap

Please have a look at these files.

020A474FED81400000000719 at 85.200.240.142 is the first call leg (in to asterisk) and 203f326067605d070c51b7f967ffa7b2 at 158.58.152.14:5060 is the second (out of asterisk).

A BYE is received from the first one, which results in a reINVITE on the second. In this case the call is actually hung up, but only because of a BYE from the callee-side.
(Restricted to Users role)
> T38 calls not being hung up correctly
> -------------------------------------
>
>                 Key: ASTERISK-26616
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26616
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.8.0
>            Reporter: Morten Tryfoss
>            Assignee: Morten Tryfoss
>         Attachments: calls.pcap, debuglog.txt, full_debug_log.txt
>
>
> We've got this scenario:
> There's a bridged call which negotiates T38. 
> After fax is sent the caller sends BYE. 
> Then Asterisk tries a reINVITE to the callee (back to normal audio). 
> This is rejected by a 488 not accepted here.
> No BYE is sent to the callee.
> The call stays up for several hours until dropped by the callee.
> I found this in the debug log at BYE:
> [Nov 21 16:35:14] DEBUG[15411][C-001906b6] chan_sip.c: Hangup call SIP/sipic2-00320075 (This is the callee-channel), SIP callid 754576b53f450b526e26caf705dbbf57 at 158.58.152.21:5060
> [Nov 21 16:35:14] DEBUG[18031][C-001906b6] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '754576b53f450b526e26caf705dbbf57 at 158.58.152.21:5060' Request 104: Found
> It seems like Asterisk is trying to CANCEL the call instead of BYE? Invitestate is incorrect  or channelstate is not up after the failed reINVITE, which in turn sets needcancel to TRUE?
> It seems to be easy to reproduce, but it takes some time for me to identify the long running calls since they're not visible in Asterisk anymore.
> The issue could also be related to ASTERISK-26609.



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