[asterisk-bugs] [JIRA] (ASTERISK-23226) chan_sip fails to transmit BYE request to WebSocket connected peer after a failed attended transfer

[Matt Jordan (JIRA)]

Matt Jordan created ASTERISK-23226:
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             Summary: chan_sip fails to transmit BYE request to WebSocket connected peer after a failed attended transfer
                 Key: ASTERISK-23226
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23226
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/WebSocket
    Affects Versions: 11.5.0
         Environment: Linux
            Reporter: Giovanni Bezicheri
            Assignee: Matt Jordan


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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