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Reported by: chewbacca
Patch by: casper
(Closes issue #11401)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90155 f38db490-d61c-443f-a65b-d21fe96a405b
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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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This makes ast_stopstream() thread-safe.
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handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
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another
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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cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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a string-based storage and comparison mechanism because it is easier to support on other platforms
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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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or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash.
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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then never turned it on (oops).
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versions of the lock routines. These are incorrect for a number of reasons:
- It breaks the build on mac.
- If there is a problem with locks not getting initialized, then the proper
fix is to find that place and fix the code so that it does get initialized.
- If additional debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS wrappers.
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88805 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88719 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88471 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: snuffy
Patch by: ys
Closes issue #11143
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@88210 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: ys
Fixed by: ys
Closes issue #11116
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is waiting for a lock, this will now show the details about who currently
has it locked.
(inspired by issue #11100)
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contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config,
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@87168 f38db490-d61c-443f-a65b-d21fe96a405b
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require the user to do that
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@87069 f38db490-d61c-443f-a65b-d21fe96a405b
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failures. If so, we could end up in infinite recursion. The only lock that
is affected by this is a mutex in astmm.c used when MALLOC_DEBUG is enabled.
(closes issue #11044)
Reported by: ys
Patches:
lock.h.diff uploaded by ys (license 281)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@86836 f38db490-d61c-443f-a65b-d21fe96a405b
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the internal mutex used to protect the lock debugging data.
(closes issue #11044, patch suggested by Ivan)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@86726 f38db490-d61c-443f-a65b-d21fe96a405b
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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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not thread safe. How ironic! Anyway, these changes ensure that the code that
is accessing the lock debugging data is thread-safe.
Many thanks to Ivan for finding and fixing the core issue here, and also
thanks to those that tested the patch and provided test results.
(closes issue #10571)
(closes issue #10886)
(closes issue #10875)
(might close some others, as well ...)
Patches: (from issue #10571)
ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229)
- a few small changes by me
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85994 f38db490-d61c-443f-a65b-d21fe96a405b
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character set don't get stripped when reading configuration.
(closes issue #10982, dandre)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85561 f38db490-d61c-443f-a65b-d21fe96a405b
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command
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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85533 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85532 f38db490-d61c-443f-a65b-d21fe96a405b
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in the middle of the struct, instead of at the end. One of the Debian folks,
paravoid, pointed out that this breaks binary compatability with modules
compiled against older headers. So, I'm moving the new member to the end
of the struct to resolve the situation.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85316 f38db490-d61c-443f-a65b-d21fe96a405b
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the caller of ast_frame_free() properly
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- Deadlock in ast_write (issue #10406)
- Deadlock in ast_read (issue #10406)
- Possible mutex initialization error in lock.h (issue #10571)
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now note the lock type for each lock that a thread holds.
(mutex, rdlock, or wrlock)
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were in the wrong place.
(closes issue #10846, alebm)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@84146 f38db490-d61c-443f-a65b-d21fe96a405b
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This was causing multiple issues, especially issue 7706 and BE-160
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@84018 f38db490-d61c-443f-a65b-d21fe96a405b
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changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@83432 f38db490-d61c-443f-a65b-d21fe96a405b
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This version of the patch maintains the original behavior of the code when
not using FastAGI.
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24)
Slight mods by me
Tested by: juggie, festr
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chan_usbradio-1.4 branch every time the configure script gets changed
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82385 f38db490-d61c-443f-a65b-d21fe96a405b
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dev mode is enabled, register the CLI command that can be used to run the astobj2
test and print out statistics.
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updated the localtime.c file from source. Next we'll have to write ast_strptime
to match.
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seeing with the addition of astobj2
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structure (context)
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you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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we don't typedef objects anywhere else in Asterisk, so we might as well make
this follow the same convention.
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everywhere since it could be a handy callback to have
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