[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:                       SWP-3479 
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 14:37 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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---------------------------------------------------------------------- 
 (0135116) lmadsen (administrator) - 2011-05-18 14:37
 https://issues.asterisk.org/view.php?id=19311#c135116 
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18
lines
  
  Allow transfer loops without allowing forwarding loops
  
  We try to avoid the situation where two phones may be forwarded to each
other
  causing an infinite loop by storing each dialed interface in a channel
  datastore and checking the list before dialing out. This works, but
currently
  breaks situations like A calls B, A transfers B to C, B transfers C to
A, and A
  transfers C to B. Since human interaction is happening here and not an
  automated forwarding loop, it should be allowed.
  
  This patch removes the dialed_interfaces datastore when a call is
bridged (a
  suggestion from the brilliant mmichelson). If a call is being bridged,
it
  should be safe to assume that we aren't stuck in a loop.
  
  Since we are now handling this is the bridge code, the previous attempts
at
  handling it in app_dial and app_queue are removed.
  
  Review: https://reviewboard.asterisk.org/r/1195/ 

Issue History 
Date Modified    Username       Field                    Change               
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2011-05-18 14:37 lmadsen        Note Added: 0135116                          
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