[asterisk-bugs] [Asterisk 0017236]: [regression] X-Lite disconnects because of RTCP timeout when Local channel and MeetMe/F involved

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17236 
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Reported By:                dimas
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   17236
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.2.9
Asterisk Version:           1.6.2.6 
JIRA:                       SWP-1335 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-04-23 11:09 CDT
Last Modified:              2010-05-06 05:46 CDT
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Summary:                    [regression] X-Lite disconnects because of RTCP
timeout when Local channel and MeetMe/F involved
Description: 
The dialplan:

context test1 {
        start => {
                Answer;
                Read(code,conf-getpin);
                MeetMe(1234,F);
        }

        test => {
                Answer;
                Dial(Local/start at test1/);
        }
}

1. use X-Lite
2. place a call to test at test1 somehow
3. you are asked for a PIN. 
4. enter anything and press #
5. You hear you are only person in the conference
6. 30 seconds later X-Lite hangs up the call because of RTCP inactivity

Note that X-Lite must have RTCP timeout enabled (this is default
setting):
  Right click -> Options -> Advanced -> Network -> 
    In times of network disruption automatically hang up the call after =>
ON
    RTCP had been inactive for => 30 seconds
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---------------------------------------------------------------------- 
 (0121460) dimas (reporter) - 2010-05-06 05:46
 https://issues.asterisk.org/view.php?id=17236#c121460 
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To be honest, I do not see how SIP trace will help. However I'm providing
it just to make sure the issue won't be closed "due to lack of feedback"
:)

Note: we are using DUNDi so I filtered these from log file using (grep -vi
dundi) - these are irrelevant to the call but produce a lot of text... 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-06 05:46 dimas          Note Added: 0121460                          
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