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

[David Woolley (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17930?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=202649#comment-202649 ] 

David Woolley commented on ASTERISK-17930:
------------------------------------------

I was originally confused because the first report appeared to be saying Asterisk is sending INVITE/Replaces, when it is actually receiving it.  This is not a simple or frequently exercised case.  In fact it was completely broken for a long time in the early 1.6s.  The sort of trace provided with the more recent report would be needed to gain any real understanding of what was happening with the first case.

In the latest report, I would suggest that this line is the key:

[Jan 31 10:14:01] DEBUG[17358] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 207af371-6a71d8ba-5c260fdb at 192.168.2.55.

I would guess that either it never actually got re-scheduled, or the channel got destroyed before it could be re-scheduled.

Incidentally, this second case needs two calls to Asterisk.  At the moment, I would have to guess that B xfer actually means that means that B sends REFER/Replaces to A and A sends INVITE/Replaces to Asterisk.
                
> 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.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators
For more information on JIRA, see: http://www.atlassian.com/software/jira