[asterisk-bugs] [JIRA] (ASTERISK-25268) Neither a srcchange or marker after (attended) transfer

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25268?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226960#comment-226960 ] 

Asterisk Team commented on ASTERISK-25268:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Neither a srcchange or marker after (attended) transfer
> -------------------------------------------------------
>
>                 Key: ASTERISK-25268
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25268
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Transfers
>    Affects Versions: 11.13.1
>         Environment: Debian/jessie
>            Reporter: dtryba
>
> On completing an attended transfer the original caller doesn't hear the transferee. Up until completion every party had a bidirectional audio.
> This is an almost duplicate of ASTERISK-15790 except here the problem is on a A leg instead of C leg. The cause is exactly the same: a huge jump in the timestamp (going backwards) without either a change of srccid or marker in rtp.
> This behavior is similar to ASTERISK-23142, it occasionally happens without using the asterisk transfer feature (using SIP UA functions instead). The Vodafone Ericsson upstream refuses to forward RTP to the endpoints (apparently with reason).
> Frequently happens with Asterisk 11.13.1~dfsg-2, the changes to channel.c are minimal (and non related) in the latest 11 LTS compared to 11.13.



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