[asterisk-bugs] [JIRA] (ASTERISK-27673) Attended SIP Transfer via func local_attended_transfer does not call TRANSFER_CONTEXT

[Peter Katzmann (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27673?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242180#comment-242180 ] 

Peter Katzmann commented on ASTERISK-27673:
-------------------------------------------

At DTMF XFER the dialplan call occurs after everything is set up and shortly before the channels are masqueraded.
This could also be achieved for SIP REFER, i played around a little bit and got it working as expected, but this was only proof of concept.
If this is working it gives me the possibility to keep track about the call flow that there was an attended transfer. 
Currently this scenario is an blind spot for me and i loose information which are crucial for our users.


> Attended SIP Transfer via func local_attended_transfer does not call TRANSFER_CONTEXT
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27673
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27673
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Transfers
>    Affects Versions: 11.25.3, 13.17.2
>         Environment: ubuntu lts 14.x, 16.x
>            Reporter: Peter Katzmann
>            Assignee: Peter Katzmann
>            Severity: Minor
>
> During tests of transfer scenarios we figured out that under some circumstances the defined TRANSFER_CONTEXT is not called. We nailed it down to Attended Transfers via SIP REFER inside a single pbx.
> Further debugging showed that inside chan_sip.c the handle_request_refer asterisk calls  local_attended_transfer. 
> If these function succeeded asterisk returns (13.x series) or jumps to cleanup (11.x series) without calling any ast_async_goto for the TRANSFER_CONTEXT variable.



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