[asterisk-bugs] [JIRA] Commented: (ASTERISK-20375) Asterisk channel reference leak when attempting to transfer a call originated to a local channel running the Echo application

[aragon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20375?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=196986#comment-196986 ] 

aragon commented on ASTERISK-20375:
-----------------------------------

Thanks David,

I've reviewed the configuration and no IVR's are used on the affected site.
All incoming lines go directly to dynamic ACD queues.
Overflow rules for each queue go to a busy tone.

But almost all outgoing calls require a PIN to be entered before a call is placed.
Also I see the Autodestructs shortly after an agent login and an agent must also enter a login password.

I've tried to reproduce this on a local system but I cannot reproduce.

I look forward to the patch ;)


> Asterisk channel reference leak when attempting to transfer a call originated to a local channel running the Echo application
> -----------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20375
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20375
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_local, Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.16.0
>            Reporter: Mark Michelson
>         Attachments: ami.pcap, asteriskcliwithwarnings.txt, gdbthreadapplyall_non_optimized.txt, ngrepami.txt, sip.pcap
>
>
> While working on a separate issue, I managed to reliably create a situation that results in a channel reference leak.
> Use the following extensions.conf:
> {noformat}
> [default]
> exten => 11,1,Answer()
> same => n,Echo()
> exten => 2301,1,Dial(Local/11 at default,,t)
> {noformat}
> 1) From the CLI issue the following command:
> {noformat}
> originate SIP/999 extension 2301 at default
> {noformat}
> where SIP/999 is a SIP phone.
> 2) Pick up SIP/999 when it starts ringing.
> 3) Press the configured attended transfer DTMF in features.conf. You will start to hear music on hold (odd).
> 4) Hang up SIP/999.
> 5) Issue a {{core show channels}} CLI command.
> You will notice that SIP/999 is still around, as are the two Local/2301 channels. Every so often, a warning message will appear
> {noformat}
> *CLI> [Sep  6 16:37:27] WARNING[610]: chan_sip.c:3918 __sip_autodestruct: Autodestruct on dialog '54200e235d36cebb6182df3d0b9ddf5f at 10.24.20.249:5060' with owner in place (Method: BYE). Rescheduling destruction for 10000 ms
> {noformat}
> Yay, you have leaking channel refs!

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