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

[Giovanni Bezicheri (JIRA)]

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

Giovanni Bezicheri updated ASTERISK-23090:
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    Description: 
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.

  was:
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 

    
> 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
>
> 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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