[asterisk-bugs] [JIRA] (ASTERISK-23090) chan_sip fails to transmit BYE request to WebSocket connected peer after a failed attended transfer

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23090?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team closed ASTERISK-23090.
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    Resolution: Suspended

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> chan_sip fails to transmit BYE request to WebSocket connected peer after a failed attended transfer
> ---------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23090
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23090
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/WebSocket
>    Affects Versions: 11.5.0
>         Environment: Linux
>            Reporter: Giovanni Bezicheri
>            Assignee: Joshua Colp
>         Attachments: issue_23090_full_log
>
>
> This bug involves the SRTP module (websocket with port 8088).
> The scenario is the following: A (normal phone), B (sipml), C (normal phone). A calls to B, B answers and a conversation is correctly estabilished between A and B. B does an attended transfer to C, C answers and hangs up. The conversation between A and B goes on. A hangs up but B does not receive the hangup event via websocket from asterisk, so it still remains in busy state.
> In the same scenario but without the transfer the hangup of A causes correctly the hangup of B (with the right communication via websocket).
> Moreover the hangup event is sent from asterisk to the standard SIP port (5060) but not via websocket.. this odd behaviour make me think about an Asterisk bug.



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