Mark with XXX one place (during command execution) where
navigation should be protected with actionlock, but is not
because it would block requests for a long time.
To solve this properly we need to put reference counts in
the struct manager_action.
A suboptimal fix is to copy the record on a search and then
unlock the list while we work on the copy.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45177 f38db490-d61c-443f-a65b-d21fe96a405b
On passing, small rearrangement of the code to reduce indentation.
There is a bit more cleanup planned for this file, so a merge to 1.4
will be done when it is all done.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45152 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006) | 7 lines
------------------------------------------------------------------------
r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006) | 2 lines
don't drop the entire permit/deny list when an attempt is made to add an invalid entry (BE-92)
------------------------------------------------------------------------
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45126 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45031 | file | 2006-10-13 11:53:22 -0400 (Fri, 13 Oct 2006) | 10 lines
Merged revisions 45030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45030 | file | 2006-10-13 11:49:53 -0400 (Fri, 13 Oct 2006) | 2 lines
Pass the right value to usleep for sleeping, and always add the background refresh item back into the scheduler if enabled since it is deleted during reload. (issue #8142 reported by p_lindheimer)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45032 f38db490-d61c-443f-a65b-d21fe96a405b
simplifications. On passing, use a single exit point.
(once done with the cleanup i will merge the changes into 1.4,
if applicable)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44626 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44379 f38db490-d61c-443f-a65b-d21fe96a405b
revert Mark's change that caused a memory leak (cap_set_proc() does not free the capability structure so we always need to call cap_free())
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44152 f38db490-d61c-443f-a65b-d21fe96a405b
- with AST_DEVMODE, building codecs/lpc10 fails because of lots
of warnings, and the configure step in editline fails as well.
Fix this by removing the -Werror in these steps.
- on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile
(a possible alternative way is to add "export AST_LIBS" near
the beginning of the file).
With this fix, i believe that some of the platform-specific
conditionals in main/Makefile are redundant (because they should
be already dealt with in the top level Makefile) but i don't
have a platform to check.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44081 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43853 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines
Fix an issue with PLAYBACKSTATUS not being set under certain circumstances.
Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string.
Fix Background() to return -1 like Playback(), if no args are specified.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43804 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43780 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43710 | russell | 2006-09-26 16:56:42 -0400 (Tue, 26 Sep 2006) | 17 lines
(This was actually BE-65)
Merged revisions 43708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | 7 lines
Back in revision 4798, this message was changed from using ast_cli() to directly
calling write(). During this change, checking if this was a remote console was
removed. This caused this message about using "exit" or "quit" to exit an
Asterisk console to come up in times where it did not make sense. This change
restores the check to see if this is a remote console before printing the
message. (fixes BE-4)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43715 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines
Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43696 f38db490-d61c-443f-a65b-d21fe96a405b
Original patch by junky, modified by anthonyl, modified again by jcolp (with minor modifications by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43300 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42600 | file | 2006-09-09 16:24:19 -0400 (Sat, 09 Sep 2006) | 2 lines
Only truly consider the channel in the same format if the format matches the raw format OR if a translation path already exists to translate between them. (issue #7887 reported by softins & issue #7803 reported by alvaro_palma_aste). Thanks goes to stubert for giving me access to a box and showing me a scenario where this occured.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42601 f38db490-d61c-443f-a65b-d21fe96a405b
for uses in cases where you *know* that it will do no good. This patch was
inspired by file for use in some work of his on mixmonitor/chanspy.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41958 f38db490-d61c-443f-a65b-d21fe96a405b
large enough and had to be reallocated, cut off the partially appended data.
Otherwise, the function will get called over and over again appending to the
end every time and never thinking it has enough room.
Thanks to jmls for access to his machine for debugging!
(issue #7691)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41944 f38db490-d61c-443f-a65b-d21fe96a405b
very stupid thing to do. It ends up duplicating the frame twice, linking in
one of them and setting the tail pointer to the other one. Sorry ...
Thanks to file for pointing out the breakage!!! file rocks.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41864 f38db490-d61c-443f-a65b-d21fe96a405b
There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41278 f38db490-d61c-443f-a65b-d21fe96a405b
when LOADABLE_MODULES is off, don't export symbols from the main binary
when LOADABLE_MODULES is off, and the compiler/linker support it, strip out code not used in the final binary
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41166 f38db490-d61c-443f-a65b-d21fe96a405b
also eliminate a warning from dlopen() because some flags were missing on the second dlopen() call
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41087 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40994 | russell | 2006-08-24 15:41:26 -0400 (Thu, 24 Aug 2006) | 11 lines
Fix a few issues related to the handling of channel variables
- in pbx_builtin_serialize_variables(), the variable list traversal would stop
on a variables with empty name/values, which is not appropriate
- When removing the GROUP variables, use AST_LIST_REMOVE_CURRENT instead of
AST_LIST_REMOVE
- During masquerading, when copying the variables list from one channel to the
other, using AST_LIST_INSERT_TAIL is not valid for appending a whole list.
It leaves the tail pointer of the list invalid. Introduce a new macro,
AST_LIST_APPEND_LIST that appends a list properly.
(issue #7802, softins)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40995 f38db490-d61c-443f-a65b-d21fe96a405b
also, keep trying to dlclose() a module until it actually goes away, since it may have other modules it brought in when it was loaded (thanks PCadach for pointing this problem out to me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40949 f38db490-d61c-443f-a65b-d21fe96a405b
allocation and free on every call of the function for preparing the string
that will be appended. Then, use the ast_dynamic_str() code instead of the
open coded version that is appended to when waiting for it to be delivered.
- use for loops for list traversals
- convert the manager sessions list to use list macros
- use atomic operations for num_sessions and usecounts
- convert some defines to the equivalent enum
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40783 f38db490-d61c-443f-a65b-d21fe96a405b
- free the string fields allocation if ast_create_channel() failes to open the
alert pipe
- formatting tweaks
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40782 f38db490-d61c-443f-a65b-d21fe96a405b
- Create an astmm_log() macro that logs the same message to both stderr as well
as the mmlog file if it is open instead of duplicating the code everywhere.
- Use for loops for list traversals instead of while loops
- reduce nesting
- ensure locking isn't put around more than is necessary
- localize a struct definition
- change the limit of the path to the mmlog to PATH_MAX instead of 80
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40780 f38db490-d61c-443f-a65b-d21fe96a405b
- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo