[asterisk-bugs] [JIRA] Commented: (ASTERISK-16912) Attended transfer failure

[Karsten Wemheuer (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-16912?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=196628#comment-196628 ] 

Karsten Wemheuer commented on ASTERISK-16912:
---------------------------------------------

The phones do not respond differently to certain conditions. The only condition where asterisk is sending two Re-Invites directly after each other is the situation with sendrpid=pai and directmedia=yes. If You disable directmedia there is only one Re-INVITE. If You use directmedia=yes and disable sendrpid there is also only one Re-INVITE. There is no problem with the phones, if there is only on Re-INVITE.

Another remark from aastra: If You use directrtpsetup=yes instead of directmedia=yes and set sendrpid=pai all is fine. In this situation there is also only one Re-INVITE.

> Attended transfer failure
> -------------------------
>
>                 Key: ASTERISK-16912
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-16912
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/Transfers
>            Reporter: Karsten Wemheuer
>         Attachments: issue-18254_directmedia.txt, issue-18254_nodirectmedia.txt, issue-att-transfer.log
>
>
> I set up an asterisk system (1.8 SVN revision 293886 checked out today) and do some testing in a SIP only environment.
> With three phones doing attended transfer I have some weird behaviour.
> phone1 calls phone2, phone2 sets call on hold and calls phone3. Then phone2 is doing an attended transfer. With Snom phones there seems to be no problem. With aastra phones the call is terminated on one side only. In the described scenario phone3 seems to be connected, whereas phone1 is idle (see issue-att-transfer.log). From Asterisk's point of view both legs are terminated.
> It doesn't matter, who is doing the transfer. If the original call (in the above senario phone1) is doing the transfer, there is also one party hung up while the other seems to be connected (on the phone side).
> This might be related to ASTERISK-16847, but the behaviour is different, so I make up a new ticket. The workaround fix from ASTERISK-16847 doesn't work (the logfile attached is from an original svn checkout, no patches applied).

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