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

Joshua Colp (JIRA) noreply at issues.asterisk.org
Tue Feb 13 06:52:13 CST 2018


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27673?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242170#comment-242170 ] 

Joshua Colp commented on ASTERISK-27673:
----------------------------------------

When the attended transfer is COMPLETED you know it is an attended transfer and can play a sound to the transferer for example. However an attended transfer is not "send this call into the dialplan at extension 1000 context test". It's "this channel is taking my place". It may go into a bridge with another channel, for example. It's not going into the dialplan in that case and TRANSFER_CONTEXT wouldn't do anything.

What would you expect the dialplan variable to do with a SIP initiated attended transfer?

> 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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