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

Matt Jordan (JIRA) noreply at issues.asterisk.org
Fri Jan 3 10:59:03 CST 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-23090?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=213568#comment-213568 ] 

Matt Jordan commented on ASTERISK-23090:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


                
> 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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