This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines
Fix a deadlock relating to channel locks and autoservice
It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of
autoservice.
The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.
The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.
In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.
It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.
(closes issue #14057)
Reported by: rtrauntvein
Patches:
14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein
Review: http://reviewboard.digium.com/r/107/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
move parsing of the qsig channel mapping configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and into a properly ifdef'd block
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
messages alerting that a channel is being ignored
if the PROC_DAHDI_NOCHAN option is set in process_dahdi.
(closes issue #13759)
Reported by: smurfix
Patches:
dahdi.patch uploaded by smurfix (license 547)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines
Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards).
(closes issue #11017)
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r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line
declared static mutexes using AST_MUTEX_DEFINE_STATIC macro
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r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line
initialize condition variable ss_thread_complete using ast_cond_init
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r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines
When creating the secondary subchannel name, it is necessary to compare to
the existing channel name without the "Zap/" or "DAHDI/" prefix, since our
test string is also without that prefix.
(closes issue #13027)
Reported by: dferrer
Patches:
chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
(Slightly modified by me, to compensate for both names)
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r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
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