ast_read to return NULL in the case that the channel has been hung up.
(crash reported by anonymouz666 on IRC in #asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a retrieval of a greeting from the database fails, but the file is found on the file system, then
we go ahead an insert the greeting into the database. The result of this is that people who
switch from file storage to ODBC storage do not need to rerecord their voicemail greetings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to the scheduler to ensure that they don't overwrite the ID of a previously
scheduled item. If there is one, it should be removed.
(closes issue #10391, closes issue #10256, probably others, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
delete classes from memory that were no longer in the config. This patch fixes
that problem as well as another one. Previously, if you reloaded MOH using the
"moh reload" CLI command, which behaved differently than "module reload ...",
MOH had to be stopped on every channel and started again immediately. However,
there was no way to tell what class was being used, so they would all fall back
to the default class.
(closes issue #10139)
Reported by: blitzrage
Patches:
asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich
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and scheduling multi-threaded. Unfortunately, we have to do some expensive
deadlock avoidance when queueing frames on to the ast_channel owner of the IAX2
pvt struct. This was already handled for regular frames, but ast_queue_hangup
and ast_queue_control were still used directly. Making these changes introduced
even more places where the IAX2 pvt struct can disappear in the context of a
function holding its lock due to calling a function that has to unlock/lock it
to avoid deadlocks. I went through and fixed all of these places to account for
this possibility.
(issue #10362, patch by me)
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ringinuse=no for the queue.
(closes issue #10215, reported by Doug, repaired by me)
Special thanks to fkasumovic for pointing out the source of the problem and to bweschke for helping to come up with a solution!
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McKeehan in issue #10184.
Upon priority change, the resource list is not NULL terminated when
moving an item to the end of the list. This makes Asterisk endlessy
loop whenever it needs to read the list. Jids with different resource and
priority values, like in Gmail's and GoogleTalk's jabber clients put
that problem in evidence.
Upon reception of a 'from' attribute with an empty resource string,
Asterisk crashes when trying to access the found->cap pointer if the
resource list for the given buddy is not empty. This situation is
perfectly valid and must be handled. The Gizmoproject's jabber client
put that problem in evidence.
Also added a few comments in the code as well as a handle for the
capabilities from Gmail's jabber client, which are stored in a caps:c tag
rather than the usual c tag.
Closes issue #10184.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: irroot
Patches:
sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
core issue. You can not call find_callno() while holding a pvt lock as this
function has to lock another (every) other pvt lock. Doing so can lead to a
classic deadlock. So, I am tracking down all of the code paths where this
can happen and fixing them.
The fix I committed earlier today was along the same theme. This patch fixes
some code down the path of authenticate_reply.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
call find_callno. You can't hold a pvt lock while calling find_callno because
it goes through and locks every single one looking for a match.
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Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks
initialized with ast_mutex_init() were not tracked. It should work now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: bhowell
Add note to sample configuration about module load order and how it can cause perfectly good queue members to be marked as invalid.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
