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

dtryba (JIRA) noreply at issues.asterisk.org
Thu Jul 23 07:06:32 CDT 2015


     [ https://issues.asterisk.org/jira/browse/ASTERISK-25268?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

dtryba updated ASTERISK-25268:
------------------------------

    Attachment: transfer.log
                extensions.conf
                sip.conf

The requested configs and a debug log of the call/transfer.

Lots of magic is happening in AGI scripts, the results are stored in variables used in extensions.

There are no SIP users, incoming calls are handled with AGI to determine authorization of the source, calls to other PBXs are made with the 
SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
scheme.

> Neither a src change 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
>            Assignee: dtryba
>         Attachments: extensions.conf, sip.conf, timestamp.pcapng.gz, transfer.log
>
>
> 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 the A leg instead of the 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, which 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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