[asterisk-bugs] [JIRA] (ASTERISK-28370) res_pjsip_t38: Not accepting Audio Re-invite after T.38 rejection

[Shane Short (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28370?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246983#comment-246983 ] 

Shane Short commented on ASTERISK-28370:
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HI All,
Has there been any progress on this issue? I can see it's being tracked internally at Digium. As I've had to completely disable T.38 on my upstream carrier, I'm not seeing a 50% fax failure rate, so ideally I'd to see if there's any workarounds I can use.

> res_pjsip_t38: Not accepting Audio Re-invite after T.38 rejection
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-28370
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28370
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_session, Resources/res_pjsip_t38
>    Affects Versions: 16.2.1
>         Environment: Asterisk Version: Asterisk 16.2.1
> Operationg System: Ubuntu 18.04.1 
> Kernel: 4.15.0-44-generic
> PJSIP Version: PJPROJECT version currently running against: 2.8
>            Reporter: Shane Short
>            Severity: Minor
>              Labels: fax, pjsip
>         Attachments: debug, pjsip.conf, sip_trace_11.txt
>
>
> When sending an audio based fax, asterisk rejects any re-invites (including back to audio after a T.38 rejection). Call flow is as follows:
> - Normal call setup using G711A
> - Upstream detects Fax Preamble and sends T.38 re-invite
> - As I have T.38 gateway disabled (due to crashes which I shall report in future), we reject the re-invite with 488.
> - The Upstream then re-invites us with G711A 
> - Asterisk rejects this re-invite with another 488. 
> - The upstream then sends a BYE.
> I've tested this behavior using the same upstream and target number on Asterisk 11 w/chan-sip, and the call flow proceeds as you would expect. 
> I have attached debug logging and a successful Asterisk 11/chan-sip Negotiation with the same upstream peer.



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