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

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27673?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242164#comment-242164 ] 

Joshua Colp commented on ASTERISK-27673:
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DTMF based attended transfer goes through Asterisk so it knows that it will end with an attended transfer and thus TRANSFER_CONTEXT can be used for the initial call. With a SIP protocol level attended transfer we have NO idea that a call is going to end up as an attended transfer until the transfer is completed, which is after the dialplan has been executed.

> 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
>            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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