[asterisk-bugs] [JIRA] (ASTERISK-17258) missing Contact header in 200 OK to INVITE

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17258?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=198526#comment-198526 ] 

Matt Jordan commented on ASTERISK-17258:
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Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions.  After testing with Asterisk 1.8, if you find this problem has not been resolved, please open a new issue against Asterisk 1.8.


                
> missing Contact header in 200 OK to INVITE
> ------------------------------------------
>
>                 Key: ASTERISK-17258
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17258
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Core/General
>            Reporter: Stanislav Kutil
>            Assignee: Matt Jordan
>         Attachments: full, missing_contact.pcap
>
>
> Several users of our softphone report a bug, that Asterisk 1.6.x does not always send Contact: header in response to our INVITE.
> The Contact arrives in 100 Trying, but not in 183 Session Progress nor 200 OK.
> We suspect is has something to do with 401 Unauthorized response to the initial INVITE.
> If there's no lag and the session goes like INVITE, 401 Unauthorized, ACK, INVITE, 100 Trying, 183 Session Progress, 200 OK, everything is fine.
> However we sometimes introduce a delay during the early stage and the session starts like this:
> > INVITE (CSeq 1)
> < 401 Unauthorized (CSeq 1) 
> > ACK (CSeq 1)
> > INVITE (CSeq 2)
> < 401 Unauthorized (CSeq 1)
> > ACK (CSeq 1)
> < 100 Trying (CSeq 2)   ..... contains Contact:
> < 183 Session Progress  .. no Contact:
> < 200 OK .. no Contact:
> we then report error as our SIP stack is unable to continue.
> ****** ADDITIONAL INFORMATION ******
> we are unsure of the exact version... I can confirm it in version 1.6.12
> here's a report from a customer:
> I upgraded this server from 1.6.2.12 to 1.6.2.15 this morning to do some further testing and now most calls are going through.  Instead of every other call origination, it is now about 9 of 10 originations are successful.

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