[asterisk-bugs] [Asterisk 0019311]: [patch] [regression] Revision 315643 app_dial breaks ring groups

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19311 
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Reported By:                mspuhler
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19311
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
JIRA:                        
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 315258 
Request Review:              
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Date Submitted:             2011-05-17 18:33 CDT
Last Modified:              2011-05-18 14:30 CDT
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Summary:                    [patch] [regression] Revision 315643 app_dial breaks
ring groups
Description: 
This is my first attempt at reporting an issue, so bear with me.  Hopefully
I can explain the issue clearly.

My problem started appearing after Revision 315643 of app_dial.  In fact,
if I put the code back in the file that was deleted, the issue is resolved.
 However, the code was deleted to address a bug.

To summarize my problem.  I have ring group 600 which contains ext 6001. 
If 6001 doesn't answer after 20 seconds, I have it set to go to another
ring group 601 which contains 6001, 6002 and 6003.  Prior to 315643, ext
6001 would continue to ring as part when calls would go to ring group 601
after ringing 600.  However, after 315643, 6001 would only ring 20 seconds
as part of ring group 600.  Extensions 6002 and 6003 would only ring as
part of ring group 601.  

The log indicates: 

[2011-05-17 16:08:03] WARNING[5047] app_dial.c: Skipping dialing interface
'SIP/6001' again since it has already been dialed


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 (0135115) lmadsen (administrator) - 2011-05-18 14:30
 https://issues.asterisk.org/view.php?id=19311#c135115 
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r318548 | twilson | 2011-05-11 13:15:39 -0400 (Wed, 11 May 2011) | 19
lines

Clean up several chan_sip reference leaks

Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.

The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.

(closes issue https://issues.asterisk.org/view.php?id=17255)
Reported by: kvveltho
Patches: 
      tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon
(license 1202)
Tested by: rgagnon, twilson, wdoekes, loloski

Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/

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Issue History 
Date Modified    Username       Field                    Change               
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2011-05-18 14:30 lmadsen        Note Added: 0135115                          
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