Commit Graph

986 Commits

Author SHA1 Message Date
Joshua Colp
8c03119ce5 If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@113296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 15:03:43 +00:00
Mark Michelson
7892ddd9f5 This fix prevents a deadlock that was experienced in chan_local. There was
deadlock prevention in place in chan_local, but it would not work in a specific
case because the channel was recursively locked. By unlocking the channel prior
to calling the generator's generate callback in ast_read_generator_actions(), we
prevent the recursive locking, and therefore the deadlock.

(closes issue #12307)
Reported by: callguy
Patches:
      12307.patch uploaded by putnopvut (license 60)
Tested by: callguy



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@113065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 16:08:45 +00:00
Joshua Colp
2cfae0a8f3 Pass in the path to Zaptel for systems that install Zaptel headers in a separate location.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04 00:52:36 +00:00
Joshua Colp
27da06a930 One thing at a time... let's get 1.4 building.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-04 00:48:01 +00:00
Dwayne M. Hubbard
0f7c06dc26 add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-03 23:57:44 +00:00
Mark Michelson
b4cf033072 Fix a race condition in the manager. It is possible that a new manager event
could be appended during a brief time when the manager is not waiting for input.
If an event comes during this period, we need to set an indicator that there is an
event pending so that the manager doesn't attempt to wait forever for an event that
already happened.

(closes issue #12354)
Reported by: bamby
Patches:
      manager_race_condition.diff uploaded by bamby (license 430)
	  (comments added by me)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02 17:36:04 +00:00
Joshua Colp
65767b4290 Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things.
(closes issue #12212)
Reported by: bamby


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 18:02:43 +00:00
Mark Michelson
f4c389df5c Initialize the __res_state structure used for dns purposes
to all 0's prior to using it. This is due to valgrind's complaints
on issue #12284 as well as an excerpt found in "Description" portion
of the online man page found here:

http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV

(pertains to issue #12284 but does not necessarily close it)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 17:21:21 +00:00
Joshua Colp
214973a574 Ensure that we do not exceed the hold's maximum size with a single frame.
(closes issue #12047)
Reported by: fabianoheringer
Tested by: fabianoheringer


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@112125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 16:45:14 +00:00
Tilghman Lesher
cb67f98a5a For FreeBSD, at least, the ifa_addr element could be NULL.
(closes issue #12300)
 Reported by: festr
 Patches: 
       acl.c.patch uploaded by festr (license 443)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@111442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-27 19:23:12 +00:00
Steve Murphy
8a02ac6f79 These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@111391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-27 13:03:28 +00:00
Jason Parker
7fbcd155c1 Put this flag back so we don't change the API.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@111280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-27 00:25:13 +00:00
Jason Parker
22078d33d4 Remove excessive smoother optimization that was causing audio glitches (small "pops")
after (about 200ms later) an "incorrectly" sized frame was received.

While it would be very nice to keep this as optimized as possible, it makes no sense
 for the smoother to be dropping random bits of audio like this.  Isn't that the
 whole point of a smoother?

Closes issue #12093.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@111245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 23:26:33 +00:00
Joshua Colp
be84adc952 Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@110628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25 14:37:35 +00:00
Russell Bryant
478de3ebb9 Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue #12266, reported by dimas)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@110395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 23:13:56 +00:00
Joshua Colp
996d3a1c2e Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
      11429-frametype.diff uploaded by qwell (license 4)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@110019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 18:20:28 +00:00
Steve Murphy
88b0da7c6c (closes issue #11442)
Reported by: tzafrir
Patches:
      11442.patch uploaded by murf (license 17)
Tested by: murf

I didn't give tzafrir very much time to test this, but if he does 
still have remaining issues, he is welcome to 
re-open this bug, and we'll do what is called for.

I reproduced the problem, and tested the fix, so I hope I
am not jumping by just going ahead and committing the fix.

The problem was with what file_save does with templates; 
firstly, it tended to print out multiple options:

[my_category](!)(templateref)

instead of 

[my_category](!,templateref)

which is fixed by this patch.


Nextly, the code to suppress output of duplicate declarations
that would occur because the reader copies inherited declarations
down the hierarchy, was not working. Thus:


 [master-template](!)
 mastervar = bar


 [template](!,master-template)
 tvar = value


 [cat](template)
 catvar = val


would be rewritten as:

 ;!
 ;! Automatically generated configuration file
 ;! Filename: experiment.conf (/etc/asterisk/experiment.conf)
 ;! Generator: Manager
 ;! Creation Date: Tue Mar 18 23:17:46 2008
 ;!
 
 [master-template](!)
 mastervar = bar

 
 [template](!,master-template)
 mastervar = bar
 tvar = value

 
 [cat](template)
 mastervar = bar
 tvar = value
 catvar = val

This has been fixed. Since the config reader 'explodes' inherited
vars into the category, users may, in certain circumstances, see
output different from what they originally entered, but it should
be both correct and equivalent.




git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 15:41:13 +00:00
Russell Bryant
1dae97f2aa Tweak spacing in a recent change because I'm very picky.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 04:06:05 +00:00
Joshua Colp
5fda7910c6 Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.
(AST-2008-002)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 14:58:39 +00:00
Mark Michelson
29a32cf040 Fix a logic flaw in the code that stores lock info which is displayed
via the "core show locks" command. The idea behind this section of code was
to remove the previous lock from the list if it was a trylock that had failed.
Unfortunately, instead of checking the status of the previous lock, we were referencing
the index immediately following the previous lock in the lock_info->locks array. 
The result of this problem, under the right circumstances, was that the lock which 
we currently in the process of attempting to acquire could "overwrite" the previous lock 
which was acquired. While this does not in any way affect typical operation, it *could*
lead to misleading "core show locks" output.




git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 22:05:49 +00:00
Jason Parker
4501c61d73 Backport revision 106439 from trunk. I didn't realize this was broken in 1.4 as well.
Closes issue #12222.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@109057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 15:15:56 +00:00
Michiel van Baak
c16da7c25b add missing break to case AST_CONTROL_SRCUPDATE
(closes issue #12228)
Reported by: andrew
Patches:
      SRC.patch uploaded by andrew (license 240)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@108961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-16 21:47:10 +00:00
Russell Bryant
efa3b46cdf Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup().  The concept is exactly like the
fixup callback that is used in the channel technology interface.  This callback
gets called when the owning channel changes due to a masquerade.  Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.

(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@108583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 21:38:16 +00:00
Russell Bryant
a169524ff6 (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/1.4@108135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 19:57:42 +00:00
Joshua Colp
4a8d87fe98 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/1.4@108083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 18:26:37 +00:00
Russell Bryant
2016f5ba10 Destroy the channel lock after the channel datastores.
(inspired by issue #12187)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@108031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 16:59:07 +00:00
Kevin P. Fleming
9f2d2cea4b check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)
(closes issue #12179)
Reported by: Netview


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:07:59 +00:00
Kevin P. Fleming
553070456b 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/1.4@107352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 11:04:29 +00:00
Tilghman Lesher
e1bccfc3fe Use non-global storage for eswitch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 21:32:24 +00:00
Russell Bryant
42caaed426 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/1.4@107161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:17:11 +00:00
Russell Bryant
0a4fc5b8c6 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/1.4@107158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:04:27 +00:00
Russell Bryant
7a47679898 Resolve a compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 17:13:17 +00:00
Russell Bryant
72e4729ad5 Fix a race condition where the generator can go away
(closes issue #12175, reported by edantie, patched by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 16:58:57 +00:00
Joshua Colp
2bf8f9fca3 Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
(closes issue #11516)
Reported by: ys
Patches:
      branch_1.4_cdr.diff uploaded by ys (license 281)
Tested by: anest, jcapp, dartvader


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 14:33:02 +00:00
Jason Parker
06ba2df183 Fix hardcoded grep in editline, were GNU grep is required.
(closes issue #12124)
Reported by: dmartin


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 22:14:45 +00:00
Joshua Colp
28c66694d3 Ignore source update control frame.
(closes issue #12168)
Reported by: plack


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 19:32:00 +00:00
Tilghman Lesher
072171ef5d Properly initialize rtp->schedid
(Closes issue #12154)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 15:20:52 +00:00
Tilghman Lesher
56e908b787 Safely use the strncat() function.
(closes issue #11958)
 Reported by: norman
 Patches: 
       20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 06:36:33 +00:00
Mark Michelson
24ca0899c2 Quell an annoying message that is likely to print every single time that
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.

Thanks to joetester on IRC for pointing this out



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-06 22:10:07 +00:00
Joshua Colp
cd703523db Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:32:10 +00:00
Russell Bryant
d564404d73 Fix a bug that I just noticed in the RTP code. The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 01:52:18 +00:00
Joshua Colp
be005c60d6 In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:10:34 +00:00
Joshua Colp
36bb1f9d46 When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:05:28 +00:00
Russell Bryant
40425a24bf Backport a minor bug fix from trunk that I found while doing random code
cleanup.  Properly break out of the loop when a context isn't found when
verify that includes are valid.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 04:31:29 +00:00
Russell Bryant
9c5af082ca Update the copyright information for autoservice. Most of the code in this file
now is stuff that I have written recently ...


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 16:01:50 +00:00
Russell Bryant
547ac9f501 Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.

1) Change it so that autoservice thread doesn't keep looping around calling
   ast_waitfor_n() on 0 channels twice a second.  Instead, use a thread condition
   so that the thread properly goes to sleep and does not wake up until a
   channel is put into autoservice.

   This actually fixes an interesting bug, as well.  If the autoservice thread
   is already running (almost always is the case), then when the thread goes
   from having 0 channels to have 1 channel to autoservice, that channel would
   have to wait for up to 1/2 of a second to have the first frame read from it.

2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
   channels and no fds to poll() on, such as was the case with the previous code
   for the autoservice thread.  In this case, the code would call alloca(0), and
   pass the result as the first argument to poll().  In this case, the 2nd
   argument to poll() specified that there were no fds, so this invalid pointer
   shouldn't actually get dereferenced, but, this code makes it explicit and
   ensures the pointers are NULL unless we have valid data to put there.

(related to issue #12116)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:50:43 +00:00
Joshua Colp
5bfdc9818d It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL.
(closes issue #11911)
Reported by: edgreenberg
Patches:
      v1-11911.patch uploaded by dimas (license 88)
Tested by: tbsky


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:28:59 +00:00
Russell Bryant
679cc09b1d Fix a major bug in autoservice. There was a race condition in the handling of
the list of channels in autoservice.  The problem was that it was possible for
a channel to get removed from autoservice and destroyed, while the autoservice
thread was still messing with the channel.  This led to memory corruption, and
caused crashes.  This explains multiple backtraces I have seen that have
references to autoservice, but do to the nature of the issue (memory corruption),
could cause crashes in a number of areas.

(fixes the crash in BE-386)
(closes issue #11694)
(closes issue #11940)

The following issues could be related.  If you are the reporter of one of these,
please update to include this fix and try again.

(potentially fixes issue #11189)
(potentially fixes issue #12107)
(potentially fixes issue #11573)
(potentially fixes issue #12008)
(potentially fixes issue #11189)
(potentially fixes issue #11993)
(potentially fixes issue #11791)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-29 23:34:32 +00:00
Russell Bryant
12e5fb358a Fix a bug in the lock tracking code that was discovered by mmichelson. The issue
is that if the lock history array was full, then the functions to mark a lock as
acquired or not would adjust the stats for whatever lock is at the end of the array,
which may not be itself.  So, do a sanity check to make sure that we're updating
lock info for the proper lock.

(This explains the bizarre stats on lock #63 in BE-396, thanks Mark!)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 22:23:05 +00:00
Jason Parker
6c3d62c1fa Make pbx_exec pass an empty string into applications, if we get NULL.
This protects against possible segfaults in applications that may try
 to use data before checking length (ast_strdupa'ing it, for example)

(closes issue #12100)
Reported by: foxfire
Patches:
      12100-nullappargs.diff uploaded by qwell (license 4)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 19:20:10 +00:00