[asterisk-bugs] [Asterisk 0017425]: Segfault after launching JACK_HOOK from AMI

[Asterisk Bug Tracker]

The following issue has been ASSIGNED. 
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https://issues.asterisk.org/view.php?id=17425 
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Reported By:                Motiejus
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   17425
Category:                   Applications/app_jack
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.2.12
Asterisk Version:           SVN 
JIRA:                       SWP-1616 
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:             2010-05-31 06:52 CDT
Last Modified:              2010-07-26 15:13 CDT
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Summary:                    Segfault after launching JACK_HOOK from AMI
Description: 
Setting JACK_HOOK channel variable from AMI leads to asterisk segfault.
However, setting JACK_HOOK from command line works OK:
When the call is started:
*CLI> dialplan set chanvar $stuff{Channel}
JACK_HOOK(manipulate,n,i(rec_$uniq:input),o(rec_$uniq:output),c(rec_$uniq))
on

works, but the same in AMI:

Action: Setvar
Channel: $stuff{Channel}
Variable:
JACK_HOOK(manipulate,n,i(rec_$uniq:input),o(rec_$uniq:output),c(rec_$uniq))
Value: on

throws segmentation fault for asterisk. Trace of connecting the call:

<snip>
[New Thread 0x2b9a16c5e910 (LWP 32621)]
    -- Executing [123456 at NPDB2:73] Monitor("SIP/1001-00000000",
"wav,myfilename") in new stack
    -- Executing [123456 at NPDB2:74] Set("SIP/1001-00000000", "DialTo=PBX2")
in new stack
    -- Executing [123456 at NPDB2:75] NoOp("SIP/1001-00000000", "PBX2") in
new stack
    -- Executing [123456 at NPDB2:76] NoOp("SIP/1001-00000000",
"SIP/1001-00000000") in new stack
    -- Executing [123456 at NPDB2:77] Dial("SIP/1001-00000000",
"SIP/PBX2/000123456,60,M(connect-jack,741586)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called PBX2/000123456
  == Begin MixMonitor Recording SIP/1001-00000000
    -- SIP/PBX2-00000001 is ringing
[New Thread 0x2b9a1745f910 (LWP 32742)]

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0x2b9a16aea910 (LWP 32607)]
0x00002b9a07d4718a in jack_activate (client=0xca23d0) at client.c:1985
1985			buf[i] = (char) (i & 0xff);
(gdb)
</snip>

It is not a problem if Jack (I suppose), because same from Dialplan and
asterisk CLI works fine.
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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-07-26 15:13 lmadsen        Asterisk Version         1.6.2.7 => SVN      
2010-07-26 15:13 lmadsen        Assigned To               => russell         
2010-07-26 15:13 lmadsen        Target Version            => 1.6.2.12        
2010-07-26 15:13 lmadsen        Description Updated                          
2010-07-26 15:13 lmadsen        Additional Information Updated                  
 
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