[asterisk-bugs] [JIRA] (ASTERISK-21486) Call answered by a dynamic agent and then SIP transferred to an external number is not written to CDR

[David Brillert (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21486?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205371#comment-205371 ] 

David Brillert commented on ASTERISK-21486:
-------------------------------------------

Summary of /var/log/asterisk/messages file attached:
The call flow comes from an external Asterisk server with phone extensions 8000 and 8001 registered.
Agent 1 is logged to extension 6000

Extension 8000 dials 5555551234 to ring queue
Agent 1 Extension 6000 answers 5555551234 and does SIP blind transfer to 8001
8001 picks up

No CDR collected for Agent 1 Extension 6000 answers 5555551234 and does SIP blind transfer to 8001

This is a serious problem because the call to extension 8001 or any other external transfer from an agent extension cannot be billed.
                
> Call answered by a dynamic agent and then SIP transferred to an external number is not written to CDR
> -----------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21486
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21486
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_queue
>    Affects Versions: 11.3.0
>         Environment: CentOS 5 64 bit Intel
>            Reporter: David Brillert
>         Attachments: Master.csv, messages
>
>
> Incoming call to app_queue is answered by an ACD agent is collected by CDR.
> The agent does SIP transfer to an external number and the CDR is not collected.
> This only effects calls to an ACD queue as far as I can tell.
> For example:
> Incoming call to an extension is collected by CDR
> if the extension does SIP transfer to an external number the CDR is collected.
> I'm attaching verbose asterisk messages log with sip set debug on and the Asterisk master.csv file for the broken scenario.

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