https://origsvn.digium.com/svn/asterisk/trunk
........
r108529 | mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 lines
Fixing a potential buffer overflow in the manager command ModuleCheck.
Though this overflow is exploitable remotely, we are NOT issuing a security
advisory for this since in order to exploit the overflow, the attacker would
have to establish an authenticated manager session AND have the system privilege.
By gaining this privilege, the attacker already has more powerful weapons at his
disposal than overflowing a buffer with a malformed manager header, so the vulnerability
in this case really lies with the authentication method that allowed the attacker to
gain the system privilege in the first place.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines
Merged revisions 108530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines
Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold. Otherwise, they just stay on like it does
when an extension is in use.
(closes issue #11263)
Reported by: russell
Patches:
notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r107721 | tilghman | 2008-03-11 15:58:42 -0500 (Tue, 11 Mar 2008) | 2 lines
Add contributed script for separation of database access from Asterisk
........
r107787 | russell | 2008-03-11 17:21:19 -0500 (Tue, 11 Mar 2008) | 6 lines
Add a trivial new dialplan function, AST_CONFIG(), which allows you to access
a variable from an Asterisk configuration file in the dialplan, or anywhere
else where dialplan functions can be used.
(Inspired by a discussion with Tilghman and Pari)
........
r107790 | russell | 2008-03-11 17:54:25 -0500 (Tue, 11 Mar 2008) | 2 lines
remove documentation of an argument that i did not implement
........
r107928 | tilghman | 2008-03-11 21:53:21 -0500 (Tue, 11 Mar 2008) | 2 lines
Cache config files, when possible, for speed
........
r108034 | russell | 2008-03-12 12:06:37 -0500 (Wed, 12 Mar 2008) | 4 lines
- Add Tilghman to the copyright info ... he wrote the hard part :)
- Remove some magic in unload_module that isn't needed. Module use counts
already ensure that the function isn't going to be in use at this point.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r107719 | russell | 2008-03-11 15:54:32 -0500 (Tue, 11 Mar 2008) | 8 lines
This patch adds support for extended help prompts in voicemail. These prompts
are in the 1.4.9 sounds release.
(closes issue #11705)
Reported by: jaroth
Patches:
helpprompts.patch uploaded by jaroth (license 50)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r106757 | murf | 2008-03-07 12:57:57 -0600 (Fri, 07 Mar 2008) | 126 lines
(closes issue #6002)
Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r108286 | kpfleming | 2008-03-12 16:37:40 -0500 (Wed, 12 Mar 2008) | 13 lines
add support for named sections in zapata.conf, and fix a few bugs in config file parsing
(closes issue #9503)
Reported by: tzafrir
Patches:
fix_cleanups uploaded by tzafrir (license 46)
zapata_sections uploaded by tzafrir (license 46)
skipchannel_options uploaded by tzafrir (license 46)
conf_sample uploaded by tzafrir (license 46)
patches updated by me to better conform to coding guidelines and fix some problems
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines
Merged revisions 108288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines
Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.
The scheduler callback will always return 0. This means that this id
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.
(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)
This is the first of potentially several such fixes.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108238 | mmichelson | 2008-03-12 16:19:30 -0500 (Wed, 12 Mar 2008) | 20 lines
Merged revisions 108227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines
Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when
it is appropriate and when it is not appropriate to use it.
I also removed the part of the debug message that mentions that this is probably a bug because
there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g.
when the scheduler callback manually reschedules with a new id instead of returning non-zero to
tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug
message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro
is probably being used in several places where it shouldn't be. Also removed the redundant line,
file, and function information since that is provided by ast_log.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines
Merged revisions 108086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines
if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP
closes issue #11475
Reported by: andrebarbosa
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108137 | russell | 2008-03-12 14:59:05 -0500 (Wed, 12 Mar 2008) | 48 lines
Merged revisions 108135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines
(closes issue #12187, reported by atis, fixed by me after some brainstorming
on the issue with mmichelson)
- Update copyright info on app_chanspy.
- Fix a race condition that caused app_chanspy to crash. The issue was that
the chanspy datastore magic that was used to ensure that spyee channels did
not disappear out from under the code did not completely solve the problem.
It was actually possible for chanspy to acquire a channel reference out of
its datastore to a channel that was in the middle of being destroyed. That
was because datastore destruction in ast_channel_free() was done near the
end. So, this left the code in app_chanspy accessing a channel that was
partially, or completely invalid because it was in the process of being free'd
by another thread. The following sort of shows the code path where the race
occurred:
=============================================================================
Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
--------------------------------------||-------------------------------------
ast_channel_free() ||
- remove channel from channel list ||
- lock/unlock the channel to ensure ||
that no references retrieved from ||
the channel list exist. ||
--------------------------------------||-------------------------------------
|| channel_spy()
- destroy some channel data || - Lock chanspy datastore
|| - Retrieve reference to channel
|| - lock channel
|| - Unlock chanspy datastore
--------------------------------------||-------------------------------------
- destroy channel datastores ||
- call chanspy datastore d'tor ||
which NULL's out the ds' || - Operate on the channel ...
reference to the channel ||
||
- free the channel ||
||
|| - unlock the channel
--------------------------------------||-------------------------------------
=============================================================================
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines
Merged revisions 108083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@108085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r107791 | tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines
An offhand comment from Russell made me realize that the configuration file
caching would not work properly for users.conf and any other file read from
more than one place. I needed to add the filename which requested the config
file to get it to work properly.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines
Merged revisions 107637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines
Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue #12136)
Reported by: aragon
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r107530 | mmichelson | 2008-03-11 10:59:32 -0500 (Tue, 11 Mar 2008) | 7 lines
Remove some redundant logic from wait_for_answer. This also let's us get rid of one of
those XXX comments from the code.
The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are
not set, so there's no need to explicitly check them again.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines
Merged revisions 107352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r107019 | murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line
way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) | 16 lines
Merged revisions 107161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines
Fix another bug specifically related to asynchronous call origination. Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.
(inspired by BE-386)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) | 17 lines
Merged revisions 107158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines
Fix some bugs related to originating calls. If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed. This caused weird memory corruption and
crashes to occur.
(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
