[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=247598#comment-247598 ] 

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

Hi joshua, i am trying to fix this issue myself and after some test i'll decided to try with asterisk 16 as you suggested in your previous post. I am afraid that in asterisk 16 things goes worst: 6004 start t38, asterisk send reinvite to 6005 with t38, 6005 refuse it, asterisk propagate back the '488 not acceptable here' to 6004; 6004 send a new invite with g711 for audio fallback; at this point asterisk 16 send another time a reinvite to 6005 with t38 and from this point things goes really wrong, asterisk respond to 6004 with SDP g711 with media port value 0, 6005 that has already refused t38, refuse it again; asterisk send a BYE to both leg and calls drop.
I would really like to fix my issue but asterisk 16 is really a mess. I really think this is a bug, can someone give it an eyes? (in attach tha call flow and the packet capture).

> 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
>            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
>
>
> 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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