[asterisk-bugs] [JIRA] (ASTERISK-23090) No websocket hangup
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Tue Jan 21 09:25:04 CST 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-23090?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-23090:
------------------------------------
Assignee: Matt Jordan (was: Giovanni Bezicheri)
Status: Triage (was: Waiting for Feedback)
> 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: Matt Jordan
> 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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