- use inaddrcmp(), since we have it
- output the type of frame and subclass being dropped, and the type/subclass that is already being processed (which caused the drop)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker
for the advice on this).
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crashes while we are trying to find a workaround.
Iksemel development seems to have stalled and we might have to stop using the
TCP/TLS connections in that library and use our own, which would scale better
from a poll/select perspective I guess. It would also make it easier to migrate
to OpenSSL and stop Asterisk from depending on both OpenSSL and GnuTLS.
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Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
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snmp library more than once without completely unloading the module and loading
it again.
(issue #9571, reported by hristo, additional helpful debug information from festr,
patch from me)
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1. VoiceMailMain was configured in the dialplan with an extension as its argument
2. A message was left for this mailbox
3. Tried to call VoiceMailMain but hung up before entering password.
This was fixed by checking that a pointer was non-null prior to trying to dereference it.
(Issue 9810, reported by xmarksthespot, patched by Corydon76 with modifications by me).
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https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
........
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this command was not locking the conference list at all.
(issue #9351, reported by and patch submitted by Junk-Y, committed patch
is different and by me)
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Every time I tried to go debug it by adding some debug output, the behavior
would change. It turns out I wasn't crazy. I had the following piece of code:
if (remove)
AST_LIST_REMOVE_CURRENT(...);
Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional
statement didn't do much good at all. It always ran at least all of the
macro minus the first statement, so I was seeing list entries magically
disappear when they weren't supposed to.
After many hours of debugging, I have come to this extremely irritating fix. :)
(issues #9581, #9497)
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unable to be played over the phone. (Issue 9786, reporter: xmarksthespot, Patched by xmarksthe spot with revisions by me,
reviewed by Russell Bryant).
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bogus on my machine. ast_safe_string_alloc() was broken. It called
vsnprintf() on a va_args list twice without re-initializing it. After the first
usage, va_end() and va_start() must be called again.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line
simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
........
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all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
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If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device.
Issue 8394, patch by DEA.
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