[asterisk-bugs] [JIRA] Feedback Entered: (ASTERISK-20284) The DIAL application does not cause asterisk to start reading packets if we have sent a "Session Progress" and the other end has sent a Re-INVITE.

[John Covert (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-20284?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

John Covert updated ASTERISK-20284:
-----------------------------------

    Send back to Developer?: I'm not done! I will comment again later to send it back.  (was: I'm done. Send it back!)
                     Status: Waiting for Feedback  (was: Waiting for Feedback)

Rusty, I've not been able to synch up with the owner of the other PBX in order to generate the more detailed data.  I've left voicemail and sent email, and hope to be able to respond very soon.

> The DIAL application does not cause asterisk to start reading packets if we have sent a "Session Progress" and the other end has sent a Re-INVITE.
> --------------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-20284
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20284
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 10.6.1, 10.7.0
>         Environment: Mac OS X 10.4.11 PowerPC
>            Reporter: John Covert
>            Assignee: John Covert
>         Attachments: ReinviteBug.txt
>
>
> If the SIP channel has sent a "Session Progress" and the other end has sent a Re-INVITE, when the call is answered in the Dial application, one-way transmission results.
> Given the following dialplan:
> [inbound]
> exten => _X!,1,Dial(SIP/x28,120,m(5xbring))
> or
> exten => _X!,1,Progress
> exten => _X!,n,Playback(pls-wait,noanswer)
> exten => _X!,n,Dial(SIP/x28)
> All works fine as long as the calling SIP endpoint does not issue a Re-INVITE.
> However, if the calling system (also Asterisk 10.7.0) issues a Re-INVITE, the calling party can hear the called party, but the called party cannot hear the caller.
> A SIP trace (attached) shows that the SIP dialogue is correct.
> The tcpdump output shows that the calling party's packets are arriving at the interface.
> An RTP trace shows that Asterisk is not reading the incoming packets.
> A unacceptable workaround is to "Answer" the call before issuing the Dial command; this, of course, is bad because it provides a false answer condition on the trunk.

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