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

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23090?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214647#comment-214647 ] 

Matt Jordan commented on ASTERISK-23090:
----------------------------------------

It does look like we are hanging up the channel coresponding with the WebRTC peer, but are not transmitting a BYE request for some reason. The place where this occurs within the log is here:

{noformat}
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] channel.c: Hanging up channel 'SIP/303-00000018'
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] chan_sip.c: Hangup call SIP/303-00000018, SIP callid 1548218469 at 192.168.5.188
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] chan_sip.c: update_call_counter(303) - decrement call limit counter on hangup
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] chan_sip.c: Updating call counter for incoming call
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] chan_sip.c: Call from peer '303' removed from call limit 2147483647
[2014-01-07 15:11:15] DEBUG[11238][C-0000000b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb4e35ef4'
[2014-01-07 15:11:15] DEBUG[6148] devicestate.c: No provider found, checking channel drivers for SIP - 303
[2014-01-07 15:11:15] DEBUG[6148] chan_sip.c: Checking device state for peer 303
[2014-01-07 15:11:15] DEBUG[6148] devicestate.c: Changing state for SIP/303 - state 1 (Not in use)
[2014-01-07 15:11:15] DEBUG[6148] devicestate.c: device 'SIP/303' state '1'
[2014-01-07 15:11:15] DEBUG[6148] devicestate.c: No provider found, checking channel drivers for SIP - 303
[2014-01-07 15:11:15] DEBUG[11182] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/303-00000018
Uniqueid: 1389103841.28
CallerIDNum: 303
CallerIDName: cristian cordless
ConnectedLineNum: 202
ConnectedLineName: cristian po
AccountCode: 303
Cause: 16
Cause-txt: Normal Clearing
{noformat}

I would expect to see a BYE transmission around this point, but one does not occur.
                
> 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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