[asterisk-bugs] [Asterisk 0013115]: could NOT get the channel lock

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13115 
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Reported By:                pj
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13115
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 132312 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-07-20 07:13 CDT
Last Modified:              2008-08-20 09:54 CDT
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Summary:                    could NOT get the channel lock
Description: 
Asterisk SVN-trunk-r132312 
I have lot of this error messages on CLI,
It seems, that it can be related to ilbc codec, because it happens mainly
when try to call echotest from xlite/eyebeam softphone with ilbc. 
When I tried same situation with gsm, it doesn't happened (or only
sporadically).


[Jul 20 14:13:17] ERROR[5827]: chan_sip.c:18837 handle_request_do: We
could NOT get the channel lock for SIP/324-083ae4e0!
[Jul 20 14:13:17] ERROR[5827]: chan_sip.c:18838 handle_request_do: SIP
transaction failed: YzcyMGFkMTgyNGNiMTRlMGFhNThkNTBkYjU0Y2FmMGY.
[Jul 20 14:13:18] ERROR[5827]: chan_sip.c:18837 handle_request_do: We
could NOT get the channel lock for SIP/324-083ae4e0!
[Jul 20 14:13:18] ERROR[5827]: chan_sip.c:18838 handle_request_do: SIP
transaction failed: YzcyMGFkMTgyNGNiMTRlMGFhNThkNTBkYjU0Y2FmMGY.

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---------------------------------------------------------------------- 
 (0091579) DennisD (reporter) - 2008-08-20 09:54
 http://bugs.digium.com/view.php?id=13115#c91579 
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I just started running Asterisk SVN-trunk-r138887M and I'm seeing the same
thing.

The debug log shows:  Failed to grab owner channel lock, trying again, in
addition to the exact same lock errors above.

In trunk's chan_sip.c the line:

copy_socket_data(&p->socket, &req->socket);

was in 1.6.0beta9:

p->socket = req->socket;

I tried to change it to that and Asterisk won't start.

This is in chan_sip.c's handle_request_do.

I get this whenever any SIP call passes through Asterisk.  It seems to
actually WORK ok (only tested 1 call at a time) and better than 1.6.0beta9
was for me. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-08-20 09:54 DennisD        Note Added: 0091579                          
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