[asterisk-bugs] [Asterisk 0017236]: 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:                
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Project:                    Asterisk
Issue ID:                   17236
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.6 
JIRA:                        
Regression:                 No 
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:              2010-04-23 12:23 CDT
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Summary:                    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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---------------------------------------------------------------------- 
 (0120839) pabelanger (manager) - 2010-04-23 12:23
 https://issues.asterisk.org/view.php?id=17236#c120839 
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We will need debug log and SIP Trace (see below).

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

---
Thank you for taking the time to report this bug and helping to make
Asterisk better. 

Unfortunately, we cannot work on this bug because your description did not
include enough information. 

You may find it helpful to read the Asterisk Issue Guidelines
http://www.asterisk.org/developers/bug-guidelines. 

We\'d be grateful if you would then provide a more complete description of
the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-04-23 12:23 pabelanger     Note Added: 0120839                          
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