[asterisk-bugs] [JIRA] (ASTERISK-28441) fax: T38 fallback to voice does not change codec

Christian Berger (JIRA) noreply at issues.asterisk.org
Fri Jul 3 09:56:25 CDT 2020


     [ https://issues.asterisk.org/jira/browse/ASTERISK-28441?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Christian Berger updated ASTERISK-28441:
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    Attachment: bug.pcap

We seem to have the same issue. We can replicate this with both 13.32.0 and 16.11.1
However in our case we always negotiated alaw and switching back to alaw after a failed attempt at T.38 doesn't work.

> fax: T38 fallback to voice does not change codec
> ------------------------------------------------
>
>                 Key: ASTERISK-28441
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28441
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_fax
>    Affects Versions: 13.26.0, 13.27.0, 16.4.1
>            Reporter: Simone Freddio
>            Assignee: Unassigned
>            Severity: Minor
>              Labels: fax, pjsip
>         Attachments: ast13-27-ata-t38-verso-ata-no-t38.cap, bug.pcap, callflow asterisk 16.4.1.png, fax_fallback_g711_ok.cap, fax with asterisk 16 filtrato.cap, t38_audio_fallback_patch
>
>
> Enviroment: Same problem on customer site, i have replicated the same issue in my lab:
> Linksys SPA112 (t38 enabled) <—> asterisk 13.27.0 <—> Linksys SPA112 (t38 disabled)
> The t38 disabled linksys mean disabled by the web interface, the pjsip endpoint still has t38_udptl=true.
> Initial call phase goes up with g729 on first and second leg, first leg (t38 enabled ata) send a reinvite to start t38, asterisk forware the invite to the t38 disabled ata that refuse it. Asterisk forward the 488 unacceptable here to first leg that after a while send a reinvite with g711 (was g729); at this point asterisk didn't forward the reinvite to the second leg that remain in g729. The call fail.
> I have also tried with FAXOPT(gateway)=yes and in this case, after 488 i have no audio at all; in 'core debug' i find:
> starting T.38 gateway for T.38 channel PJSIP/dev6004-00000007 and G.711 channel PJSIP/dev6005-00000008
> but channel PJSIP/dev6005 is still in g729! After a while i have a lot of:
> DEBUG[5497][C-00000001] translate.c: Sample size different 160 vs 1280



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