[asterisk-bugs] [JIRA] Feedback Requested: (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.

[Rusty Newton (JIRA)]

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

Rusty Newton updated ASTERISK-20284:
------------------------------------

    Assignee: John Covert
      Status: Waiting for Feedback  (was: Triage)

John, can provide an asterisk log with the VERBOSE and DEBUG message types enabled (level 5 at least)? If you can also provide the full pcap from tcpdump as well that would be helpful.

> 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.

--
This message is automatically generated by JIRA.
For more information on JIRA, see: http://www.atlassian.com/software/jira