[asterisk-bugs] [Asterisk 0018818]: Crashing when using local channels and realtime on asterisk 1.8.3-rc2

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=18818 
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Reported By:                nic
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18818
Category:                   Channels/chan_local
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.3-rc2 
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:             2011-02-15 14:13 CST
Last Modified:              2011-02-16 16:28 CST
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Summary:                    Crashing when using local channels and realtime on
asterisk 1.8.3-rc2
Description: 
I have been having a problem with asterisk crashing when using local
channels and realtime on asterisk 1.8.3-rc2.
The example given here is I think the easiest way to reproduce this
problem.

In extensions.conf I have:

[internal]
switch => Realtime/extensions/p
exten => 301,1,Answer()
exten => 301,2,Dial(Local/501 at internal)
exten => 301,3,Hangup()
exten => 501,1,Answer()
exten => 501,2,Playback(demo-echotest)
exten => 501,3,Echo()
exten => 501,4,Hangup()

Where dialling 301 puts you through to 501 and you hear the echo test
message fine. However if I move 501 to the realtime database extensions
table and remove it from extensions.conf asterisk hangs on the local
channel dial, then completely dies a few minutes later (console stops
responding to commands) with killall -9 asterisk being the only way to stop
it.

In both cases I can dial 501 directly with no problem.

The last message on the console (with verbose 10) -- Executing
[300 at internal:2] Dial("SIP/1014-00000001", "Local/501 at internal")

Everything works fine with the exact same setup and asterisk 1.8.1.2 and
1.8.2.3.

Discussed a little on asterisk-users, Jonathan Thurman managed to
reproduce the problem with latest SVN of 1.8 branch.
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Issue History 
Date Modified    Username       Field                    Change               
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2011-02-16 16:28 lmadsen        Description Updated                          
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