No subject

....
https://issues.asterisk.org/view.php?id=8  0x00007fa2f66b552f in
sip_set_rtp_peer (chan=0xd8d998,
instance=0xd9c278, vinstance=0x0, tinstance=0x0, codecs=7437, nat_active=0)
at chan_sip.c:27693

  line 27649: sip_pvt_lock(p);    // lock is being set

https://issues.asterisk.org/view.php?id=7  0x00007fa2f665888d in
transmit_reinvite_with_sdp (p=0xd85f88,
t38version=0, oldsdp=0) at chan_sip.c:11020
https://issues.asterisk.org/view.php?id=6  0x00007fa2f66424c5 in
try_suggested_sip_codec (p=0xd85f88) at
chan_sip.c:5996
https://issues.asterisk.org/view.php?id=5  0x0000000000519fc0 in
pbx_builtin_getvar_helper (chan=0xd8d998,
name=0x7fa2f66c81c3 "SIP_CODEC_INBOUND") at pbx.c:9467

  line: 9467: ast_channel_lock(chan);   // lock is set again and waiting
for the first lock????

https://issues.asterisk.org/view.php?id=4  0x0000000000444ac0 in __ao2_lock
(user_data=0xd8d998, file=0x5b8a54
"pbx.c", func=0x5beb50 "pbx_builtin_getvar_helper", line=9467, var=0x5bd339
"chan") at astobj2.c:157
....

I assume my patch is more like a workaround, possibly introducing new bugs
(by temporary unlocking it)
====================================================================== 

---------------------------------------------------------------------- 
 (0134994) alecdavis (manager) - 2011-05-16 17:02
 https://issues.asterisk.org/view.php?id=19274#c134994 
---------------------------------------------------------------------- 
If it is the same, from https://issues.asterisk.org/view.php?id=18837 use
bug18837-trunk.diff3.txt

The actual diff for 1.8svn that was applied is at
http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?r1=308679&r2=308945&pathrev=308945


Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-16 17:02 alecdavis      Note Added: 0134994                          
======================================================================