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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26616?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233852#comment-233852 ] 

Asterisk Team commented on ASTERISK-26616:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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
>         Attachments: debuglog.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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