[asterisk-bugs] [JIRA] (ASTERISK-27949) res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers

[George Joseph (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27949?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

George Joseph updated ASTERISK-27949:
-------------------------------------

    Assignee: Ross Beer  (was: George Joseph)
      Status: Waiting for Feedback  (was: Open)

Are the phones expecting the specific string "Call completed elsewhere"?

I ask because we only send a "SIP" protocol reason code if the reason code is ANSWERED_ELSEWHERE so if we suppress the Q.850 Reason, then the phones won't get any reason at all for reasons other than ANSWERED_ELSEWHERE.

To fix this, I wanted to use the ast_str2cause() function to generate a "SIP" Reason header for all reason codes BUT it returns simply "Answered elsewhere" for the ANSWERED_ELSEWHERE code and I can't change that.  Also is getting more SIP reason headers going to confuse the phones further?

If we don't generate the extra SIP reason codes, should we suppress the Q.850 response ONLY for ASWERED_ELSEWHERE?  I.E.  Should the option be "suppress_q850_reasons" or "suppress_q850_answered_elsewhere_reason"?




> res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers
> --------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27949
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27949
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_rfc3326
>    Affects Versions: 13.21.1, 15.4.1
>         Environment: CentOS 7
>            Reporter: Ross Beer
>            Assignee: Ross Beer
>
> A lot of endpoints such as Cisco, Grandstream and Yealink have an issue with multiple 'Reason' headers.
> Removing the second Q.850 Reason header resolves the 'Answered Elsewhere' issue on all.
> It's not possible to remove the Q.850 header totally as this is used to pass hangup reasons between asterisk servers and other SIP services.
> Therefore a configurable option per endpoint would resolve this issue.



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