[asterisk-bugs] [JIRA] (ASTERISK-17930) Attended transfer - transfering phone left connected

[Pawel Sternal (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17930?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=202428#comment-202428 ] 

Pawel Sternal commented on ASTERISK-17930:
------------------------------------------

I have a similiar scenario. A call to B, B call to Asterisk, B xfer, Asterisk receives INVITE packet with Replaces, A and Asterisk stay connected, but B holding B to Asterisk. In attached there're [^invite_replaces] with log when Asterisk received INVITE with Replaces.
                
> Attended transfer - transfering phone left connected
> ----------------------------------------------------
>
>                 Key: ASTERISK-17930
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17930
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/Transfers
>            Reporter: Maciej Krajewski
>            Assignee: Maciej Krajewski
>            Severity: Minor
>         Attachments: invite_replaces, mk.pcap
>
>
> The issue 0015833 still exists in newest version of Asterisk.
> When doing a remote attended transfer in one of these 2 setups:
> phones A,B,C --- proxy --- asterisks Z,X
> when A->B call is on Z and B->C is on X, or:
> phones A,B (with identity B1,B2), C --- asterisks Z,X
> (A,B1 register on Z; B2,C on X)
> when A->B1 call is on Z and B2->C is on X
> In both scenarios Z and X are friends with no authentication needed.
> The B phone doesn't get properly disconnected. asterisks invite/replace each other properly and the audio channel is ok. B itself drops one of the calls. But Z is not disconnecting B's call at all. You can replicate that scenario with minimalistic dialplan - _X.,Dial(SIP/${EXTEN}) in default on both sides.

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