[asterisk-bugs] [Asterisk 0018371]: [patch] asterisk crash when dialing SIP/${var} where var is empty or not set

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18371 
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Reported By:                gbour
Assigned To:                qwell
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Project:                    Asterisk
Issue ID:                   18371
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Target Version:             1.4.41
Asterisk Version:           SVN 
JIRA:                       SWP-2652 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 295998 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-11-24 08:58 CST
Last Modified:              2011-01-31 17:07 CST
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Summary:                    [patch] asterisk crash when dialing SIP/${var} where
var is empty or not set
Description: 
exten = 500,1,NoOp()
exten = 500,n,Set(var=)
exten = 500,n,Dial(SIP/${var})

if var is not set or empy, executing Dial(SIP/${var}) make asterisk crash
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 (0131317) svnbot (reporter) - 2011-01-31 17:07
 https://issues.asterisk.org/view.php?id=18371#c131317 
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Repository: asterisk
Revision: 305254

_U  branches/1.8/
U   branches/1.8/apps/app_dial.c
U   branches/1.8/channels/chan_sip.c

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r305254 | qwell | 2011-01-31 17:07:01 -0600 (Mon, 31 Jan 2011) | 24 lines

Merged revisions 305253 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
  r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17
lines
  
  Merged revisions 305252 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10
lines
    
    Prevent a crash when dialing a technology with no destination (ex:
Dial(SIP/))
    
    chan_iax2 and other channel drivers already had code to prevent this. 
The
    attempt that app_dial was making to prevent it was not correct, so I
fixed that.
    
    (closes issue https://issues.asterisk.org/view.php?id=18371)
    Reported by: gbour
    Patches: 
          18371.patch uploaded by gbour (license 1162)
  ........
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http://svn.digium.com/view/asterisk?view=rev&revision=305254 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-31 17:07 svnbot         Checkin                                      
2011-01-31 17:07 svnbot         Note Added: 0131317                          
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