[asterisk-bugs] [Asterisk 0011386]: Asterisk 1.4.14 and load average.

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Jan 10 14:25:28 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11386 
====================================================================== 
Reported By:                flujan
Assigned To:                Corydon76
====================================================================== 
Project:                    Asterisk
Issue ID:                   11386
Category:                   Core-General
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.14  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             11-27-2007 08:11 CST
Last Modified:              01-10-2008 14:25 CST
====================================================================== 
Summary:                    Asterisk 1.4.14 and load average.
Description: 
Asterisk starts do use 500% of CPU and incresease the load average of the
machine to 61. Generating also a core dump.


Bt output and valgrind.txt.


====================================================================== 

---------------------------------------------------------------------- 
 andrew - 01-10-08 14:25  
---------------------------------------------------------------------- 
This 1.4 branch update (97077) to chan_sip.c causes hold problems with
polycom and grandstream phones. I was going to open a new bug (as it's a
new problem), but I'll stick it here for now. If I back off chan_sip then
it works again.

It does not seem to affect cisco or sipura phones I have for testing.

When I try to place a call on hold, the grandstream just disconnects it's
side of the call but asterisk does not disconnect the other side. The
polycom puts the call on hold on the phone, but asterisk does not see it as
on hold. The polycom can't resume the call so it's stuck. If it's
disconnected then the phone still sees it on hold. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-10-08 14:25  andrew         Note Added: 0076684                          
======================================================================




More information about the asterisk-bugs mailing list