[asterisk-bugs] [JIRA] (ASTERISK-29253) Incorrect bridging on transfer

[Igor Olhovskiy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29253?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258802#comment-258802 ] 

Igor Olhovskiy commented on ASTERISK-29253:
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[~jcolp], just to say, I really appreciate your (and for sure all the community) work that you're doing! And pretty aware of how busy could be core developers here and why this kind of stuff can take so much time to resolve.
That's exactly why I proposed my assistance here to follow your guidelines, that is how community should work from my point of view - everybody is doing their best, so I can help the whole community to profit from Yury's patch. 
I think remarks like "non-interested" was referred to some maybe not super successful negotiations for commercial support, which are more money, than community questions.
So, again - many thanks for job you're doing despite it's your day-off!

> Incorrect bridging on transfer
> ------------------------------
>
>                 Key: ASTERISK-29253
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29253
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_simple
>    Affects Versions: 16.15.0, 18.1.1
>         Environment: Ubuntu Linux 18.04.5 LTS
>            Reporter: Yury Kirsanov
>            Assignee: Unassigned
>         Attachments: bridge_simple.tar.gz, bridge_softmix.tar.gz, call_flow.txt
>
>
>  We have an Asterisk server and one SIP device registered with it and one SIP trunk to another system. Also we another SIP device to call for test purposes.
> Here's a simplified diagram of call flow with attended transfer we're trying to achieve:
> SIP Device A -> Asterisk PBX -> SIP Trunk -> External User -> Attended transfer -> Asterisk PBX -> SIP Device B.
> SIP Device A originates a call to some pre-defined number that's routed into SIP trunk (TLS+SRTP). Remote party behind SIP trunk ("External user") answers this call and then starts attended transfer to SIP Device B on Asterisk PBX. That SIP trunk uses Re-INVITE with Replaces header in order to complete transfer. During transfer SIP Device A can hear MOH after External User initiated attended transfer to SIP Device B. Then External User tries to complete the transfer connecting SIP Device A with SIP Device B. Call connects but SIP Device A continues to hear Music On Hold while SIP Device B can hear what SIP Device A says.
> Now, if we unload module 'bridge_simple' Asterisk PBX starts to use 'bridge_softmix' module and connect calls correctly, SIP Device A can establish two way communication with SIP Device B. But during attended transfer no MOH is played at all even though Asterisk shows messages like 'Starting music on hold'. And without 'bridge_simple' no Music On Hold is played at all even if we set up just a local extension that plays MOH, like this:
> exten=>100,1,Answer()
> exten=>100,n,MusicOnHold()
> If we load bridge_simple then MOH is played fine but during transfer calls are bridged incorrectly.
> I'm happy to provide full logs upon request as I don't want to edit them.
> Also, if we use SIP REFER method for transferring calls there's another issue - when External User tries to finalize call transfer Asterisk drops established call to SIP Device B and immediately re-dials it. I believe this happens because SIP Trunk is not passing Replaces in Refer-To header, but that's another issue.



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