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

[Joshua C. Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29253?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp updated ASTERISK-29253:
--------------------------------------

    Assignee: Yury Kirsanov
      Status: Waiting for Feedback  (was: Triage)

You haven't specified which SIP implementation is in use, as well please do sanitize logs and attach them. Attaching them means that any individual is able to investigate, see, and potentially resolve. By not sanitizing you limit yourself to just the members of the Sangoma Asterisk team unless someone takes interest and directly coordinates for log information. We leave this as a last resort as a result, and highly prefer attached logs.

As well, is anything behind NAT? Have you configured Asterisk to be behind NAT? What environment is this on? Virtualized? Do you have a timing module active? Have you verified everything without involving transfers?

I ask because some of your statements seem as though re-INVITEs for direct media may be going on.

> 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, 17.9.0, 17.9.1, 18.1.0, 18.1.1
>         Environment: Ubuntu Linux 18.04.5 LTS
>            Reporter: Yury Kirsanov
>            Assignee: Yury Kirsanov
>            Severity: Minor
>
> 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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