The ast_set_callerid() function needed to lock the channel. Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.
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executed in the dialplan if you have debug set to anything non-zero. This seems pointless
due to the fact that these channel variables are not referenced anywhere else in the code and
their names are esoteric enough that they would not be practical to reference in the dialplan. Plus
the fact that this behavior isn't documented anywhere means that the change is not likely to cause
any disruption. If anything, this may actually cause a slight performance increase if running with
debug on.
The motivating influence for this code change is the eventwhencalled option for queues. If set to
vars, all channel variables will be output to the manager. These unnecessary channel variables make
the output a lot more difficult to deal with.
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This set of changes fixes an issue that was reported to me on IRC yesterday.
The user, d1mas, was using chan_zap for incoming calls and was having DTMF
recognition issues in some situations. Specifically, he noticed that the
problem occurred when using DISA or WaitExten. He also noticed that when
using Read, the problem did not occur. His system also used DUNDi for
dialplan lookups.
So, he theorized that if the DUNDi lookups blocked for some period of time,
that audio from the zap channel could get lost. If the audio got lost, then
it wouldn't be run through the DTMF detector, and digits could get lost.
He was correct, and the following set of changes fixes the problem. However,
the changes go a little bit further than what was necessary to fix this exact
problem.
1) I updated pbx_extension_helper() to autoservice the associated channel to
handle cases where extension lookups may take a long time. This would
normally be a dialplan switch that does some lookup over the network, such
as the DUNDi or IAX2 switches.
This ensures that even while a DUNDi lookup is blocking, the channel will be
continuously serviced.
2) I made a change to the autoservice code. This is actually something that
has bothered me for a long time. When a channel is in autoservice, _all_
frames get thrown away. However, some frames really shouldn't be thrown
away. The most notable examples are signalling (CONTROL) frames, and DTMF.
So, this patch queues up important frames while a channel is in autoservice.
When autoservice is stopped on the channel, the queued up frames get stuck
back on the channel so that they can get processed instead of thrown away.
3) I made another change to the autoservice code to handle the case where
autoservice is started on channels recursively.
Previously, you could call ast_autoservice_start() multiple times on a
channel, and it would stop the first time ast_autoservice_stop() gets
called. Now, it will ensure that autoservice doesn't actually stop until
the final call to ast_autoservice_stop().
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codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
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invalid, due to the repetition of certain parameters in a single event.
This caused various issues for XML parsers, some of which refused to parse
at all, given the invalidity of the rendered XML. So this commit fixes
the XML output, ensuring that each entity parameter has a unique name, thus
ensuring valid XML.
Reported by: msetim
Patch by: tilghman
(Closes issue #10220)
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the conlock as well as the hints lock, it must be locked in that respective order.
In order to prevent a potential deadlock, we need to lock the conlock prior to
locking the hints lock in ast_hint_state_changed (see the call stack example on
issue #11323 for how this can happen).
(closes issue #11323, reported by eelcob, suggestion for patch by eelcob, patch by me)
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what build options were used. We agreed that we should remove this before
making a 1.4 release, and then we can put it back in. Then, we can take a
month or so to play around with it to get it how we want it.
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If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer.
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ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
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Previously, the SRV record support in Asterisk was broken. There was no
guarantee on what record Asterisk would choose to actually use. This set of
changes improves the situation by ensuring that Asterisk will choose the
highest priority record.
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The issue here is that the channel frame readq handling got broken when the
code was converted to use the linked list macros. It caused corruption of the
list head and tail pointers. So, I fixed up the usage of the linked list
macros and in passing, simplified the code. I also documented what the code
is doing, as it was a bit difficult to figure out at first.
This bug showed itself with crashes showing messed up head/tail pointers for
the readq. However, there are a couple of crashes that aren't quite as obvious,
but I think may be related. So, if your bug gets closed by this commit, but
you still have a problem, please reopen or create a new bug report.
(closes issue #10936)
(closes issue #10595)
(closes issue #10368)
(closes issue #11084)
(closes issue #10040)
(closes issue #10840)
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any channel datastores from the old channel to the new one. However, it did
not use the linked list macros properly to accomplish the task. The existing
code would only work if there was only a single datastore on the old channel.
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and ast_string_field_free_all to ast_string_field_reset_all
to avoid misuse (due to too similar names and an error in
documentation). Fix two related memory leaks in app_meetme.
No need to merge to trunk, different fix already applied there.
Not applicable to 1.2
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when traversing the list of allocated chunks so that they can be printed out
to the CLI.
(Thanks to eliel on #asterisk-dev for pointing this out!)
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the BEGIN is less than that of the defined minimum DTMF duration.
(closes issue #11051)
Reported by: casper
Patches:
channel.c.86664.diff uploaded by casper (license 55)
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and modify channel data that may change elsewhere. I went through every timer
callback in the source tree to make sure that none of them did any additional
locking that could introduce deadlocks, and all is well.
(closes issue #10765)
Reported by: Ivan
Patches:
ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229)
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me in issue #11018, doesn't really make sense. There is no reason to have
the base64 decode function force a '\0' terminated buffer, when the result is
almost always binary, anyway. In fact, this caused some breakage, as some code
in res_crypto passed in a buffer exactly the right size to get its binary
result, which got stomped on by this patch.
(closes issue #11018, reported by dimas)
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some amount of time. Be a little bit more careful and prepare all of the
output in an intermediary buffer while holding a global resource. Then, after
releasing it, send the output to ast_cli().
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and retrieve its output. The issue was that there was no way for the main Asterisk
process to know that the remote console was connecting in the -rx mode. The way that
James has fixed this is to have all remote consoles muted by default. Then, regular
remote consoles automatically execute a CLI command to unmute themselves when they
first start up.
(closes issue #10847)
Reported by: atis
Patches:
asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176)
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CLI command at once for a remote console.
(closes issue #10888)
Reported by: jamesgolovich
Patches:
asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176)
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that when events are appended to the master event queue, they use the number
of active sessions as a use count so it will know when all active sessions
at the time the event happened have consumed it. However, the handling of
the number of sessions was not properly synchronized, so the use count was
not always correct, causing an event to disappear early, or get stuck in
the event queue for forever.
(closes issue #9238, reported by bweschke, patch from Ivan, modified by me)
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Tilghman Lesher
Russell Bryant
Kevin P. Fleming
Mark Michelson
Olle Johansson
Steve Murphy
Joshua Colp
Jason Parker
Luigi Rizzo
Dwayne M. Hubbard
Doug Bailey