[asterisk-bugs] [JIRA] (ASTERISK-22558) Asterisk should not send re-invite and update during P-Asserted in call

[Steve Davies (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22558?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=218192#comment-218192 ] 

Steve Davies commented on ASTERISK-22558:
-----------------------------------------

No solution yet, but definitely repeatable. Worse than that, I have 2 cases, one which causes it and one which does not, but I still cannot see what is causing the differing behaviour!

Same for both cases:
- Inbound SIP call
- Answer() + Playback(...)
- Caller DTMF press causes Goto
- Set CONNECTEDLINE(num)
... CoLP Re-INVITE sent
- Set CONNECTEDLINE(name)
... CoLP UPDATE sent because we are already in-dialog

Continues with working case:
- Dial() a SIP device to connect caller to.
... Onward INVITE sent to new channel
... CoLP UPDATE 2 sent because we are already in-dialog
... 100 Trying received for INVITE
... 500 Error received for UPDATE
... 500 Error received for UPDATE 2
... OK/ACK proceeds for Re-INVITE, call is okay.

or with not working case:
- Start a Playback(...)
... 100 Trying received for INVITE
... 500 Error received for UPDATE
... OK received for Re-INVITE
... Asterisk never sends ACK, call fails

It is possible to work around the differences and cause it to work reliably by either using the 'i' parameter when setting CONNECTEDLINE, or by setting 'disallowed_methods=UPDATE' in sip.conf


> Asterisk should not send re-invite and update during P-Asserted in call
> -----------------------------------------------------------------------
>
>                 Key: ASTERISK-22558
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22558
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/CallerID
>    Affects Versions: 11.5.1
>            Reporter: David Brillert
>            Assignee: David Brillert
>         Attachments: call_ID_CS2.pcapng, sip.conf.txt
>
>
> An incoming call to Asterisk IVR and user dials digits to dial extension.
> The trunk has PAI yes the extension has PAI yes.
> Asterisk sends the PAI update and re-invite in same call then the SBC responds with a 500 internal server error and Asterisk never sends an ACK.
> pcap attached.  There is no RTP and the dial plan continues to execute.



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