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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27673?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242161#comment-242161 ] 

Asterisk Team commented on ASTERISK-27673:
------------------------------------------

The module you are reporting the issue against is no longer supported as a core module but your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available.

Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1]

If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way.

[1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
[2]: http://www.asterisk.org/community/discuss
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

> 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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