Commit Graph

1591 Commits

Author SHA1 Message Date
Tilghman Lesher
e6fc9ae52c Add a linkedlist macro that maintains a sorted list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:19:31 +00:00
Jason Parker
dd2700d0b1 Only try to detect silence when we actually need to, instead of...always.
If this is wrong, I'd love to hear why.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:16:31 +00:00
Jason Parker
6412a96e43 Large cleanup of DSP code
Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.

2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.

3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.

4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.

5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.


(closes issue #11968)
Reported by: dimas
Patches:
      v2-11968-dsp.patch uploaded by dimas (license 88)
      v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:05:51 +00:00
Tilghman Lesher
ef4eff9a9b Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 18:39:06 +00:00
Mark Michelson
43d70915bb This ensures that the manager interface is not enabled by default. Prior to this
change, it was possible to start Asterisk with the manager interface enabled, then
either comment out the enabled option or make manager.conf unopenable and the manager
interface would still be enabled.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 00:02:31 +00:00
Joshua Colp
738e4ec94e Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25 15:18:41 +00:00
Joshua Colp
358ac2f76a Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

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/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25 14:39:45 +00:00
Joshua Colp
30d85b3144 Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21 17:58:59 +00:00
Russell Bryant
6430ec3294 Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

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/trunk@110396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 23:14:13 +00:00
Russell Bryant
4e72f83d3e Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue #12164, reported by milazzo and jsmith, patch by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 20:08:26 +00:00
Mark Michelson
ff9befa36a Add missing unlock
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 18:01:36 +00:00
Russell Bryant
bccebdd21f Remove astobj.h from some places where it wasn't needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:45:29 +00:00
Russell Bryant
3c6cf5dcc5 Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue #12164, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
 - I moved a single line so that the sample queue update happened before
   ast_write().  The reason that this was a bug is that the G.722 frame
   originally says it has 320 samples in it (which is correct).  However,
   when the frame is written to a channel that uses RTP, main/rtp.c modifies
   the frame to cut the number of samples in half before it sends it on
   the wire.  This is to account for the stupid incorrect G.722 spec that
   makes it so we have to lie about the number of samples with RTP.  I should
   probably go and re-work the RTP code so it doesn't modify the frame so
   that a bug like this won't happen in the future.  However, this change to
   MOH is harmless.

main/channel.c:
 - I made two fixes in regards to generator timing.  Generators use samples
   for timing.  However, this code assumed 8 kHz samples.  In one case, it was
   a hard coded 160 samples, that is now written as the sample rate / 50.  The
   other place was dealing with timing a generator based on frames coming from
   the other direction.  However, that would have only worked if the sample
   rates for the formats in both directions were the same.  The code now takes
   into account that the sample rates may differ, and scales the generator
   samples accordingly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:41:22 +00:00
Jason Parker
9e3603dac9 Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue #11968


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 22:25:34 +00:00
Jason Parker
8d4276578a Rename very poorly named function to reflect what it actually does. This was causing quite a bit of confusion for me...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 21:56:15 +00:00
Joshua Colp
3e439e9616 Merged revisions 110019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

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/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 18:25:33 +00:00
Joshua Colp
e097cc7221 Add the ability to use a pattern match for a hint.
(closes issue #7767)
Reported by: Corydon76
Patches:
      20070314__simple_hint_lookup.diff.txt uploaded by Corydon76
      pbx-trunk-98436.diff uploaded by plack (license 365)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 16:54:12 +00:00
Steve Murphy
14e1d8c6d8 Merged revisions 109908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines

(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/trunk@109942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 16:24:51 +00:00
Kevin P. Fleming
75cb5032e6 actually implement HTTP request dispatching based on both URI and method; reduce duplication of data when generating responses using ast_http_error()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 16:18:29 +00:00
Russell Bryant
4c6486782f Fix some more breakage that I introduced when changing extension state callbacks to the list macros.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 15:45:49 +00:00
Kevin P. Fleming
84b133bd81 clean up code to conform to coding guidelines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 15:41:54 +00:00
Russell Bryant
89ad4ace67 Remove an unneeded variable. This compiled, but I missed the uninitialized warning
because I always compile without optimizations turned on.  Sorry!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 15:22:13 +00:00
Russell Bryant
b47eee2187 Convert handling of extension state callbacks to the list macros.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 04:32:13 +00:00
Russell Bryant
e1bd198bc0 Minor coding style changes, including adding handling for memory allocation failure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 04:14:12 +00:00
Russell Bryant
f1d2a11aad Minor change to use Asterisk macros
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 04:09:55 +00:00
Russell Bryant
c1cf92d304 Merged revisions 109838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) | 2 lines

Tweak spacing in a recent change because I'm very picky.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 04:06:31 +00:00
Kevin P. Fleming
e191b51a08 start the process of changing HTTP request dispatching to do it based on *both* URI and method, so that POST support can move into a module; move http.c's private function prototypes into _private.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 22:32:26 +00:00
Terry Wilson
b02bc230af Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se.  I also added format attributes to any printf wrapper functions I found that didn't have them.  -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:43:34 +00:00
Joshua Colp
760fc3403c Make sure values are interpreted as character strings and not format strings.
(AST-2008-004)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:13:07 +00:00
Joshua Colp
10cdbe28a8 Merged revisions 109386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines

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/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:08:09 +00:00
Terry Wilson
e727d15d34 Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory.
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.

If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 22:10:06 +00:00
Mark Michelson
b246e010d5 Merged revisions 109226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines

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/trunk@109227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 22:06:44 +00:00
Steve Murphy
0af58d3f5c (closes issue #12238)
Reported by: mvanbaak
Tested by: murf, mvanbaak

Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context
that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the
old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the
includes, ignorepats, and switches. I added code to do this immediately after the context is created.

This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also 
in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and
therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates
into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely
order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we 
merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any
elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully,
this is pretty rare thing.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 17:47:36 +00:00
Michiel van Baak
b311134430 Merged revisions 108961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines

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/trunk@108962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-16 21:50:58 +00:00
Russell Bryant
072eb8a913 Remove a double write lock of the contexts lock in ast_wrlock_contexts().
How did this ever work?

(closes issue #12219)
Reported by: ys
Patches: 
      pbx.c.diff uploaded by ys (license 281)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-15 16:21:04 +00:00
Mark Michelson
963a2cec51 Make this compile
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 21:47:55 +00:00
Russell Bryant
835df7d30f Merged revisions 108583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines

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/trunk@108584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 21:40:43 +00:00
Mark Michelson
d236e3d1b1 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/trunk@108529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 20:59:00 +00:00
Russell Bryant
c3c56990df Make the default prefix empty, like it was in Asterisk 1.4.
(closes issue #12198, reported by bkruse, patched by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 22:49:26 +00:00
Russell Bryant
8bbef5f996 Rename ast_tcptls_server_instance to session_instance, since this pertains to
server and client usage.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 22:13:18 +00:00
Joshua Colp
a3c7b08d19 Doxygenify slinfactory a bit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 21:06:50 +00:00
Russell Bryant
b38cb44acd 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
  --------------------------------------||-------------------------------------
  =============================================================================

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 19:59:05 +00:00
Joshua Colp
5fc569f5f5 Merged revisions 108083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 18:29:33 +00:00
Russell Bryant
f7e28b12fe Merged revisions 108031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) | 4 lines

Destroy the channel lock after the channel datastores.

(inspired by issue #12187)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 17:02:57 +00:00
Tilghman Lesher
10609251f9 Revert several changes from revision 102525, as the changes were not
compatible, and, in fact, introduced regressions.
(Closes issue #12190)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 05:46:39 +00:00
Tilghman Lesher
d02f74ebfe 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/trunk@107791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 22:55:16 +00:00
Joshua Colp
b84cdbfe38 Merged revisions 107646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 lines

Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted.
(closes issue #11707)
Reported by: iam

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 19:23:28 +00:00
Joshua Colp
f26ed3f4bf Clarify comment about masquerading and playback of the parking slot.
(closes issue #12180)
Reported by: davidw


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 15:05:17 +00:00
Kevin P. Fleming
c7eebb3db8 Merged revisions 107408 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines

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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:09:49 +00:00
Kevin P. Fleming
79c3038ee5 Merged revisions 107352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 11:36:51 +00:00