[asterisk-bugs] [JIRA] (ASTERISK-23090) No websocket hangup

Giovanni Bezicheri (JIRA) noreply at issues.asterisk.org
Tue Jan 7 08:25:03 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-23090?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=213689#comment-213689 ] 

Giovanni Bezicheri edited comment on ASTERISK-23090 at 1/7/14 8:24 AM:
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I attached the full log of the operations involving this issue. Referring to the description of the issue, in the log case A is 303, B is 202 and C is 203.
                
      was (Author: gbezicheri):
    This is the full log of the operations involving this issue. Referring to the description of the issue, in this case A is 303, B is 202 and C is 203.
                  
> No websocket hangup
> -------------------
>
>                 Key: ASTERISK-23090
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23090
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 11.5.0
>         Environment: Linux
>            Reporter: Giovanni Bezicheri
>            Assignee: Giovanni Bezicheri
>         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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