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

[Jon Wright (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29253?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258792#comment-258792 ] 

Jon Wright commented on ASTERISK-29253:
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Yury - I've been struggling with this exact same problem with my deployments. We're a large college group over in the UK using multiple asterisk deployments hooked into Nortel and Mitel telephone exchanges and the attended transfers (which are continually used by our switchboard staff and are "external" to asterisk) always result in the one-way MOH / incorrect bridging. I've also tried unloading 'bridge_simple' to work around the problem, but in our case it breaks the refers so we cant get the transfers to go through at all.

Josh - I imagine there's a lot more people out there affected by this issue than has been reported. I've been struggling with it for months and only now realised it was a bridging bug when stumbled across Yury's issue.

Really keen to get some traction behind a fix.

Kind Regards,

Jon

> 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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