[asterisk-bugs] [Asterisk 0010406]: [patch] Asterisk stops processing calls

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10406 
====================================================================== 
Reported By:                callguy
Assigned To:                Corydon76
====================================================================== 
Project:                    Asterisk
Issue ID:                   10406
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-08-2007 11:44 CDT
Last Modified:              10-10-2007 06:09 CDT
====================================================================== 
Summary:                    [patch] Asterisk stops processing calls
Description: 
Asterisk stops processing calls in the sip channel at random intervals.
This appears to be related to reloading chan_sip.so. When this happens the
console partially locks up (show channels becomes unresponsive) and shortly
after sip processing ceases to function. 

Running bt attached. 
====================================================================== 

---------------------------------------------------------------------- 
 Ted Brown - 10-10-07 06:09  
---------------------------------------------------------------------- 
I'm testing now latest svn branch (85199). When I make a SIP call to a
queue, THERE IS NO AUDIO when the agent answers it. Besides, i've seen the
following behaviors in transfers A->B->C to queue (A=linksys spa941, B=X,
C=linksys spa941):

X=eyebeam 1.5.16.1:   Attended transfer: crash
                      Blind transfer: no audio A<-->C
X=zoiper 2.09(latest) Attended transfer: crash
                      Blind transfer: no audio A<-->C
X=linksys spa941      Attended transfer: no audio A<-->C
                      Blind transfer: no audio A<-->C

All transfers, doing the same but calling directly to the C extension,
work fine.

This behavior is caused by latest patch, as I've tested svn 85057 with 
the patch and i saw the same. Without it, only attended transfers with
eyebeam would crash

You can use any of the configurations I have already uploaded in bug 10809
to reproduce this

The truth is i'm very surprised nobody had noticed the no audio issue... 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
10-10-07 06:09  Ted Brown      Note Added: 0071752                          
======================================================================