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

[Simone Freddio (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28441?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247602#comment-247602 ] 

Simone Freddio commented on ASTERISK-28441:
-------------------------------------------

Hi Joshua, i get it working on 13.21-cert3... can be a starting point...

> 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, 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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