[asterisk-bugs] [Asterisk 0016337]: [patch] Segmentation Fault on Originate command.

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Dec 2 09:50:30 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16337 
====================================================================== 
Reported By:                Parantido
Assigned To:                dvossel
====================================================================== 
Project:                    Asterisk
Issue ID:                   16337
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.1.10 
JIRA:                       SWP-455 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-11-27 10:11 CST
Last Modified:              2009-12-02 09:50 CST
====================================================================== 
Summary:                    [patch] Segmentation Fault on Originate command.
Description: 
Hi,

I set up a lab in this way:

PSTN <---> Nortel Meridian Option11C <--- PRI Trunk ---> Asterisk 1.6.1.10
<----> SIP CLIENTS

Sip clients are able to successfully dial External PSTN Number or Nortel
Meridian Extension Number.

But I'm experiencing weird behavior when i'm trying to issue an originate
command over DAHDI channels (i discovered this during application
development) with the following syntax:

*CLI> originate DAHDI/G1/302 extension xxxxx at outbound_route

or by Manager in this way (carriage return & new line was reported only
for completeness):

Action: Originate\r\n
Channel: DAHDI/G1/302\r\n
Context: outbound_route\r\n
Exten: 302\r\n
Priority: 1\r\n
Timeout: 30000\r\n\r\n

Asterisk output looks like this:

*CLI> originate DAHDI/G1/302 extension 3482513178 at outbound_route
    -- Requested transfer capability: 0x00 - SPEECH
    -- Executing [302 at from-pstn:1] NoOp("DAHDI/30-1", "PSTN ==> Incoming
call for: 302 from ") in new stack
    -- Executing [302 at from-pstn:2] Goto("DAHDI/30-1", "users,400,1") in
new stack
    -- Goto (users,400,1)
    -- Executing [400 at users:1] NoOp("DAHDI/30-1", "Users ==> Incoming call
for: 400 from ") in new stack
    -- Executing [400 at users:2] Set("DAHDI/30-1", "DNDVAL=") in new stack
    -- Executing [400 at users:3] NoOp("DAHDI/30-1", "Users ==> Extension 400
DND: ") in new stack
    -- Executing [400 at users:4] GotoIf("DAHDI/30-1", "0?maxreached") in new
stack
    -- Executing [400 at users:5] Set("DAHDI/30-1", "GROUP()") in new stack
Segmentation fault (core dumped)
monkeyvoice:~#

I think is a DAHDI related issue because if I use following command:

*CLI>  originate SIP/400 extension xxxxxxx at outbound_route
    == Using SIP RTP CoS mark 5
    -- Executing [xxxxxxx at outbound_route:1] NoOp("SIP/400-00000000",
"Default OutGoing Rule for: 3482513178") in new stack
    -- Executing [xxxxxxx at outbound_route:2] Dial("SIP/400-00000000",
"DAHDI/G1/xxxxxxx") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called G1/xxxxxxx
    -- DAHDI/31-1 is proceeding passing it to SIP/400-00000000
    -- DAHDI/31-1 is ringing
    -- DAHDI/31-1 answered SIP/400-00000000
    -- Hungup 'DAHDI/31-1'
  == Spawn extension (outbound_route, xxxxxxx, 2) exited non-zero on
'SIP/400-00000000'

Asterisk is able to successfully complete call setup.
====================================================================== 

---------------------------------------------------------------------- 
 (0114519) svnbot (reporter) - 2009-12-02 09:50
 https://issues.asterisk.org/view.php?id=16337#c114519 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 232269

_U  trunk/
U   trunk/funcs/func_groupcount.c

------------------------------------------------------------------------
r232269 | dvossel | 2009-12-02 09:50:30 -0600 (Wed, 02 Dec 2009) | 15
lines

Merged revisions 232268 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9
lines
  
  fixes segfault in func_groupcount
  
  closes issue https://issues.asterisk.org/view.php?id=16337)
  Reported by: Parantido
  Patches:
        issue_16337.diff uploaded by dvossel (license 671)
  	  Tested by: Parantido, dvossel
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=232269 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-02 09:50 svnbot         Checkin                                      
2009-12-02 09:50 svnbot         Note Added: 0114519                          
======================================================================




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