[asterisk-bugs] [JIRA] (ASTERISK-26869) res_pjsip: blind call transfer w/o a user name doesn't go to the s extension

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26869?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235777#comment-235777 ] 

Asterisk Team commented on ASTERISK-26869:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> res_pjsip: blind call transfer w/o a user name doesn't go to the s extension
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-26869
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26869
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_refer
>    Affects Versions: 13.14.0
>            Reporter: Torrey Searle
>
> When doing a call transfer to a sip uri w/o a user name the transfer fails instead of entering the dialplan



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