[asterisk-bugs] [Asterisk 0012494]: asterisk locks after p2p sip channel bridge

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Jun 24 06:08:50 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12494 
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Reported By:                pj
Assigned To:                murf
====================================================================== 
Project:                    Asterisk
Issue ID:                   12494
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 114536 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             04-22-2008 13:22 CDT
Last Modified:              06-24-2008 06:08 CDT
====================================================================== 
Summary:                    asterisk locks after p2p sip channel bridge
Description: 
simple call between two sip phones (both have same codec), 
console log and 'core show locks' attached
this bug was probably caused after huge commits in rev 114190,
it happens in 100% of sip calls, when p2p bridge is attempted, 
so it's really big issue!
====================================================================== 

---------------------------------------------------------------------- 
 pj - 06-24-08 06:08  
---------------------------------------------------------------------- 
I tried fresh trunk r124539, it doesn't crash anymore, 
but issue with peer not matching peer definition in sip.conf (as I wrote
in this bugreport msg id http://bugs.digium.com/view.php?id=85907) still
persist. Peer is successfully
registered and qualified OK, but when peer try to dial, asterisk doesn't
match peer entry as defined in sip.conf. As consequence, call is placed
from context defined in [general] instead of peer context and also other
options from peer's entry is not used/rewrited (like caller id). I'm using
xlite softphone, same issue has my colegue, with twinkle softphone. 
If you need another/detailed debug, please let me know.

[Jun 24 09:42:32] Using INVITE request as basis request -
YzA4MmRkYTAxYmI2Mjc4MmYwYjZlNDQ5NGRiODA4MWI.
[Jun 24 09:42:32] No user '324' in SIP users list
[Jun 24 09:42:32] No matching peer for '324' from '193.85.164.154:10838' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-24-08 06:08  pj             Note Added: 0089140                          
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