[asterisk-bugs] [Asterisk 0017167]: RECONNECT fails to work for Conference Call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17167 
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Reported By:                Starbug
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17167
Category:                   PBX/General
Reproducibility:            random
Severity:                   major
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:             2010-04-11 22:07 CDT
Last Modified:              2010-04-12 08:55 CDT
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Summary:                    RECONNECT fails to work for Conference Call
Description: 
Hello,

this is my first post, so please help me put it together right.

We are developing on Asterisks using AMI and have noticed that from time
to time RECONNECT command fails to properly work, dropping the call on
hold

Here are the steps to reproduce the problem:
1. 3000 (Zoiper) make call to 3001 (3CX)
2. 3001 Answer
3. 3001 Init Conference to 3002  (X-lite)
4. 3002 Answer
5. 3001 Complete Conference
6. 3002 leave from conference
7. 3001 Init Conference to 3002   * Zoiper hangup but CTI Client 3000
still talking
8. 3002 Answer

I am enclosing both working and not-working logs in "additional
information" section.

I would really appreciate someone get back to me on this, because I expect
a lot more people to be affected by this.
 

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 (0120265) pabelanger (manager) - 2010-04-12 08:55
 https://issues.asterisk.org/view.php?id=17167#c120265 
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Welcome, a few things about your posting.

1. Please do not post logs into the 'Additional Information' field, simply
upload them to the issue as an attachment. 

2. Your steps to reproduce information is good, but you forgot to mention
which technology you are using (SIP, IAX2, DADHI).

3. The log file you have posted is not from Asterisk, it looks to be from
http://asterisk-java.org/. You have 2 options, a) seek support from their
tracker (asterisk-java.org). or b) send the proper asterisk logs for us to
debug (see below).

--
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 didn't include enough information. You may find it helpful to
read "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               
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2010-04-12 08:55 pabelanger     Note Added: 0120265                          
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