[asterisk-commits] file: branch mattf/asterisk-ss7 r39539 - in /team/mattf/asterisk-ss7: ./ apps...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Fri Aug 11 12:18:52 MST 2006


Author: file
Date: Fri Aug 11 14:18:51 2006
New Revision: 39539

URL: http://svn.digium.com/view/asterisk?rev=39539&view=rev
Log:
Merged revisions 37809,37829-37830,37857,37881,37896,37907,37925,37936-37937,37950-37953,37973-37974,37988,37999-38000,38009-38010,38021,38030,38042,38051,38060,38068,38076-38078,38088-38089,38099,38108,38117-38118,38127,38129,38139,38148-38149,38158,38168,38187,38201,38223,38235,38250,38258,38271,38283,38286,38291,38311,38329-38330,38348,38368-38369,38371,38374,38389,38418,38421-38422,38439,38441,38452,38454,38464-38466,38476,38489,38502,38521,38532-38533,38548,38573,38586-38587,38612,38633,38652,38655,38687,38714,38732,38762,38785,38801,38823,38826,38853,38871,38887,38905,38929,38951,38973,38994,39014,39032,39057-39058,39082,39109,39131-39132,39151,39171,39182,39203,39206,39229,39248,39272,39274,39295,39316,39332-39333,39349,39351,39353,39378,39380,39404,39430-39431,39455,39479,39500,39512-39514,39516 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r37809 | tilghman | 2006-07-17 18:39:02 -0400 (Mon, 17 Jul 2006) | 9 lines

Blocked revisions 37808 via svnmerge

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r37808 | tilghman | 2006-07-17 17:36:56 -0500 (Mon, 17 Jul 2006) | 2 lines

Backport buffer increase to 1.2

........

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r37829 | russell | 2006-07-17 19:26:22 -0400 (Mon, 17 Jul 2006) | 10 lines

Blocked revisions 37828 via svnmerge

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r37828 | russell | 2006-07-17 19:25:33 -0400 (Mon, 17 Jul 2006) | 3 lines

if asked to duplicate a frame that has no data, don't set the frame's data
pointer past the end of the allocatted buffer for the new frame

........

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r37830 | russell | 2006-07-17 19:31:24 -0400 (Mon, 17 Jul 2006) | 3 lines

if asked to duplicate a frame that has no data, don't set the frame's data
pointer past the end of the buffer allocated for the new frame

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r37857 | russell | 2006-07-17 20:43:57 -0400 (Mon, 17 Jul 2006) | 10 lines

Merged revisions 37856 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r37856 | russell | 2006-07-17 20:41:47 -0400 (Mon, 17 Jul 2006) | 2 lines

don't crash if the frame has no data, but has a src

........

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r37881 | file | 2006-07-18 12:22:26 -0400 (Tue, 18 Jul 2006) | 2 lines

Expand speech API so that the developer can interact with the engine more directly and use specific functions of the connector even if a generic API call is not available

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r37896 | markster | 2006-07-18 16:26:38 -0400 (Tue, 18 Jul 2006) | 2 lines

Eliminate file descriptor leak in built in HTTP server

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r37907 | rizzo | 2006-07-18 17:19:59 -0400 (Tue, 18 Jul 2006) | 3 lines

move variable declarations to the beginning of a block.


................
r37925 | murf | 2006-07-18 22:55:24 -0400 (Tue, 18 Jul 2006) | 1 line

committing a fix for colons in goto arguments and suppressing operator messages in globals assignments.
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r37936 | markster | 2006-07-19 09:28:38 -0400 (Wed, 19 Jul 2006) | 2 lines

First pass at in-place file manipulation via manager

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r37937 | kpfleming | 2006-07-19 09:50:19 -0400 (Wed, 19 Jul 2006) | 2 lines

revert this morning's incorrect indentation change

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r37950 | kpfleming | 2006-07-19 13:16:08 -0400 (Wed, 19 Jul 2006) | 2 lines

restore buildability (bad marko!)

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r37951 | kpfleming | 2006-07-19 13:16:25 -0400 (Wed, 19 Jul 2006) | 10 lines

Merged revisions 37949 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r37949 | kpfleming | 2006-07-19 12:10:10 -0500 (Wed, 19 Jul 2006) | 2 lines

ensure that global 'maxauthreq' is reset to zero during 'reload'

........

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r37952 | kpfleming | 2006-07-19 13:21:33 -0400 (Wed, 19 Jul 2006) | 2 lines

restore buildability

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r37953 | markster | 2006-07-19 13:26:47 -0400 (Wed, 19 Jul 2006) | 2 lines

Fix some reinitialization of prev!

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r37973 | kpfleming | 2006-07-19 14:33:14 -0400 (Wed, 19 Jul 2006) | 2 lines

use new sound file version

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r37974 | kpfleming | 2006-07-19 14:50:55 -0400 (Wed, 19 Jul 2006) | 2 lines

make installation of French and Spanish sounds work properly

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r37988 | kpfleming | 2006-07-19 16:44:39 -0400 (Wed, 19 Jul 2006) | 2 lines

merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)

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r37999 | russell | 2006-07-19 17:07:14 -0400 (Wed, 19 Jul 2006) | 2 lines

add notes on the changes to music on hold handling

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r38000 | russell | 2006-07-19 17:52:59 -0400 (Wed, 19 Jul 2006) | 2 lines

add menuselect_stub.c to deps list

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r38009 | north | 2006-07-19 18:40:36 -0400 (Wed, 19 Jul 2006) | 2 lines

Fix a few doxygen warnings.

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r38010 | north | 2006-07-19 18:48:00 -0400 (Wed, 19 Jul 2006) | 2 lines

Remove "initialization from incompatible pointer type" warnings.

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r38021 | russell | 2006-07-20 12:18:07 -0400 (Thu, 20 Jul 2006) | 4 lines

add a verbose message to the AGI command, STREAM FILE, similar to the
verbose messages when using Playback, Background, or the GET DATA command
(issue #7297, softins)

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r38030 | file | 2006-07-20 14:42:22 -0400 (Thu, 20 Jul 2006) | 2 lines

Only bitmaskify the RTP payload structure for video if an RTP structure exists for it... otherwise the default values will cause codec combination madness

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r38042 | russell | 2006-07-21 13:31:28 -0400 (Fri, 21 Jul 2006) | 4 lines

Merge a new implementation of ast_inet_ntoa, our thread safe replacement for
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.

................
r38051 | bweschke | 2006-07-21 15:10:58 -0400 (Fri, 21 Jul 2006) | 3 lines

 This corrects the crash condition present in #7575, but I'm not really sure if it's the "right" fix. Please review and make any adjustments you see necessary.  


................
r38060 | russell | 2006-07-21 18:36:25 -0400 (Fri, 21 Jul 2006) | 3 lines

make ast_state2str thread safe by using thread local storage instead of a
static buffer for storing the result when the state value is unknown

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r38068 | russell | 2006-07-21 19:58:14 -0400 (Fri, 21 Jul 2006) | 2 lines

slightly simplify a loop and replace an open coded version of ast_skip_blanks

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r38076 | russell | 2006-07-21 20:08:21 -0400 (Fri, 21 Jul 2006) | 9 lines

remove an XXX comment and document that ast_autoservice_start() will return -1
if the channel is already in the autoservice list.

Why is this a valid case to return -1, you ask?  Well, there should never be
any code where it is not clear if the channel is in autoservice or not because
trying to read frames from a channel that is in the autoservice list will lead
to bad results because more than one thread will be waiting on frames to arrive
on the channel and then trying to read them.

................
r38077 | russell | 2006-07-21 20:33:56 -0400 (Fri, 21 Jul 2006) | 5 lines

use ast_atomic_fetchadd_int in chan_zap, sip, and iax2 for usecount handling
and fix a couple little things in passing
 - usecnt was not initialized in chan_iax2
 - ast_update_use_count() was not called after incrementing the count in chan_sip

................
r38078 | russell | 2006-07-21 20:38:30 -0400 (Fri, 21 Jul 2006) | 2 lines

ast_pbx_outgoing_cdr_failed() is not used outside of pbx.c so make it static

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r38088 | russell | 2006-07-22 22:41:02 -0400 (Sat, 22 Jul 2006) | 8 lines

various cleanups regarding coding guidelines issues
 - malloc to ast_malloc
 - malloc + memset to ast_calloc
 - sizeof(struct foo) to sizeof(*bar)
 - remove indentation of the entire body of a function by returning immediately
   on an allocation failure
(issue #7581, tempest1)

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r38089 | russell | 2006-07-22 22:50:25 -0400 (Sat, 22 Jul 2006) | 2 lines

update to mention svn instead of cvs

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r38099 | russell | 2006-07-22 23:29:19 -0400 (Sat, 22 Jul 2006) | 2 lines

remove a stray newline

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r38108 | russell | 2006-07-23 00:58:56 -0400 (Sun, 23 Jul 2006) | 4 lines

take the advice of an XXX comment and use an atomic operation to decrement a
variable.  Also, change a use of strdup to ast_strdup and remove a duplicated
error message.

................
r38117 | russell | 2006-07-23 01:06:47 -0400 (Sun, 23 Jul 2006) | 3 lines

resolve another XXX comment by implementing proper handling of control frames
in ast_write(), which is to call the channel's indicate function if it exists

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r38118 | russell | 2006-07-23 01:20:59 -0400 (Sun, 23 Jul 2006) | 3 lines

fix a seg fault in this application if no context paramater is given
(issue #7571)

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r38127 | russell | 2006-07-23 11:19:16 -0400 (Sun, 23 Jul 2006) | 16 lines

Merge team/russell/ast_cli_tls into the trunk.

This improves the performance of ast_cli() by not doing a heap memory
allocation and deallocation every single time the function is called. Instead,
a thread-specific buffer is allocatted the first time the function is called
and automatically free'd when the thread exits. Also note that this buffer will
only be allocatted in threads that actually call this function, which is 
probably only the threads spawned to service connected asterisk consoles.

This does introduce a new limitation on the maximum length of the resulting
string from the arguments passed to ast_cli. Previously there was no limit
since it was just allocating a buffer big enough every time the function was
called. The current buffer size is 16kB. If there is ever a case where we want
to print more than 16k characters in a single call to ast_cli(), this will have
to be increased.

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r38129 | russell | 2006-07-23 11:21:44 -0400 (Sun, 23 Jul 2006) | 4 lines

ast_malloc is sufficient here, since snprintf is called on the buffer
immediately after it gets allocated, which will automatically NULL terminate
the string

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r38139 | file | 2006-07-23 23:42:27 -0400 (Sun, 23 Jul 2006) | 2 lines

Only deal with getting the supported payloads on audio if an audio RTP stream exists

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r38148 | rizzo | 2006-07-24 07:45:06 -0400 (Mon, 24 Jul 2006) | 3 lines

fix uninitialized variable


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r38149 | rizzo | 2006-07-24 07:46:27 -0400 (Mon, 24 Jul 2006) | 3 lines

move middle-of-the-block declaration at the top.


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r38158 | file | 2006-07-24 11:47:59 -0400 (Mon, 24 Jul 2006) | 2 lines

Start out with cleared RTP payload structures instead of defaults. This should prevent issues where if a stream (audio/stream) is not present and it's RTP payload structure is combined with the overall capability then the capability would be every codec that Asterisk supports.

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r38168 | murf | 2006-07-24 13:24:47 -0400 (Mon, 24 Jul 2006) | 5 lines

blocked rev 38167 on the 1.2 branch from trunk, because it's
coded differently on trunk, would conflict and have to be hand-merged
anyway.


................
r38187 | murf | 2006-07-24 16:58:27 -0400 (Mon, 24 Jul 2006) | 7 lines

Added a small fix to exclude the AMD optimized assembly file from being included in the compile.
I don't particular like the approach, which basically tries to exclude the file based on
NOT being in a list of arches. This, IMHO, is backwards, it should be a list of AMD arches 
instead. "Better to light a candle, than to curse the darkness", is apropo here. But since I don't
have an exhaustive list of what those arches could be, I leave as-is for now.


................
r38201 | russell | 2006-07-25 15:51:31 -0400 (Tue, 25 Jul 2006) | 18 lines

This exact deadlock situation that I observed can't happen in trunk due to the
recent hold changes so that MOH is not started on the bridged channel directly.
However, the change is still not a bad idea.

Merged revisions 38200 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38200 | russell | 2006-07-25 15:43:38 -0400 (Tue, 25 Jul 2006) | 6 lines

This resolves a deadlock that a tech support customer was getting frequently
when his users would answer call waiting. If another thread is currently
holding the zt_pvt lock for the first channel, unlock both channels and let
asterisk retry the native bridge, just like what is done for the second channel
directly below these changes.

........

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r38223 | russell | 2006-07-26 03:48:48 -0400 (Wed, 26 Jul 2006) | 5 lines

merge the changes from my ast_cli_tls branch ...

Instead of having a static buffer size of 16k, start out with 256 bytes and
increase the size of the buffer as needed.

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r38235 | file | 2006-07-26 11:27:45 -0400 (Wed, 26 Jul 2006) | 10 lines

Merged revisions 38234 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38234 | file | 2006-07-26 11:26:06 -0400 (Wed, 26 Jul 2006) | 2 lines

Put default callerid into contact when the one specified is either NULL or has a zero string length. (issue #7590 reported by key2)

........

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r38250 | tilghman | 2006-07-26 13:43:17 -0400 (Wed, 26 Jul 2006) | 2 lines

Only unlock these if they were locked on entry

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r38258 | tilghman | 2006-07-26 15:59:25 -0400 (Wed, 26 Jul 2006) | 2 lines

Typo found on the -dev list

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r38271 | russell | 2006-07-26 19:12:47 -0400 (Wed, 26 Jul 2006) | 4 lines

fix setting the CFLAGS for building codec libs so that they are built with
astmm support and astmm doesn't get really upset and complain that it is being
asked to free memory that was never allocated

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r38283 | russell | 2006-07-26 21:29:57 -0400 (Wed, 26 Jul 2006) | 3 lines

fix some breakage of ast_cli() that resulted in seg faults on Josh's machine.
I'm not sure why this never caused problems for me ...

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r38286 | russell | 2006-07-26 21:49:57 -0400 (Wed, 26 Jul 2006) | 3 lines

fix crashes with MALLOC_DEBUG enabled that were a result of my recent thread
storage changes (fixes issue #7595)

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r38291 | russell | 2006-07-26 22:00:20 -0400 (Wed, 26 Jul 2006) | 11 lines

Merged revisions 38288 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38288 | russell | 2006-07-26 21:58:41 -0400 (Wed, 26 Jul 2006) | 3 lines

fix a crash when MALLOC_DEBUG is enabled and the regexten is enabled. The crash
would occur when the extension got removed. (fixes issue #7484)

........

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r38311 | kpfleming | 2006-07-26 22:46:22 -0400 (Wed, 26 Jul 2006) | 10 lines

Merged revisions 38310 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38310 | kpfleming | 2006-07-26 21:43:49 -0500 (Wed, 26 Jul 2006) | 2 lines

don't do useless translation destroy/build when the channel is already in the correct format

........

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r38329 | russell | 2006-07-27 00:28:09 -0400 (Thu, 27 Jul 2006) | 12 lines

Merged revisions 38328 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38328 | russell | 2006-07-27 00:25:41 -0400 (Thu, 27 Jul 2006) | 4 lines

Fix crash when using the "regexten" option with MALLOC_DEBUG enabled.  This was
not reported in the bug tracker but the same bug has been demonstrated in other
places in the code.

........

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r38330 | russell | 2006-07-27 00:44:49 -0400 (Thu, 27 Jul 2006) | 3 lines

fix seg fault when the parked call that timed out was the last one in the list
of parked calls (fixes issue #7565)

................
r38348 | kpfleming | 2006-07-27 11:42:49 -0400 (Thu, 27 Jul 2006) | 10 lines

Merged revisions 38347 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38347 | kpfleming | 2006-07-27 10:40:03 -0500 (Thu, 27 Jul 2006) | 2 lines

do a better job avoiding translation path teardown/setup when not needed

........

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r38368 | kpfleming | 2006-07-27 18:20:52 -0400 (Thu, 27 Jul 2006) | 8 lines

various cleanups:
  use API call for finding channel by name prefix
  code formatting to match guidelines (lost about half the of the indenting)
  remove useless automatic variable initializations
  don't set the spying channel's read format to SLINEAR when we don't do anything with the voice frames we read from it anyway
  use proper option argument checking for volume argument


................
r38369 | kpfleming | 2006-07-27 18:21:46 -0400 (Thu, 27 Jul 2006) | 2 lines

restore infinite timeout for ast_waitfor() call

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r38371 | kpfleming | 2006-07-27 18:24:19 -0400 (Thu, 27 Jul 2006) | 2 lines

block this patch from 1.2, it's already fixed here as part of a larger commit

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r38374 | kpfleming | 2006-07-27 19:00:27 -0400 (Thu, 27 Jul 2006) | 3 lines

remove local channel finding wrappers
move guts of dialplan application into separate function, so it can be shared bythe new application i'm about to add :-)

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r38389 | kpfleming | 2006-07-27 19:16:08 -0400 (Thu, 27 Jul 2006) | 4 lines

more simplification, and correct a bug i introduced in the last commit
fix prototype for a channel walking function to use a const input pointer
use existing channel walk by name prefix instead of reproducing that code in this app

................
r38418 | kpfleming | 2006-07-28 14:03:57 -0400 (Fri, 28 Jul 2006) | 4 lines

more slinfactory structure definition to inside implementation module
make read/write/hold work on samples, not bytes
add an API call to find out how many samples are available in a slinfactory

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r38421 | file | 2006-07-28 14:53:43 -0400 (Fri, 28 Jul 2006) | 10 lines

Merged revisions 38420 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38420 | file | 2006-07-28 14:49:00 -0400 (Fri, 28 Jul 2006) | 2 lines

Make a copy of the request URI in check_user_full instead of modifying the one on the structure, and also strip params properly from the user portion of the SIP URI so as to preserve the domain (issue #7552 reported by dan42)

........

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r38422 | kpfleming | 2006-07-28 14:59:59 -0400 (Fri, 28 Jul 2006) | 3 lines

move slinfactory structure definition back to header... it's just easier to use this way
add infrastructure for whispering onto a channel

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r38439 | russell | 2006-07-28 15:17:56 -0400 (Fri, 28 Jul 2006) | 8 lines

As discussed and decided on the asterisk-dev mailing list ...
- Fix some breakage I introduced a while ago that made the timestamps option
  not functional for CLI verbose output.
- Remove the use of the timestamps option for log output, since it was not
  functional.
- clarify text referring to the timestamps option so that it is clear that it
  only applies to CLI verbose output

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r38441 | kpfleming | 2006-07-28 15:58:53 -0400 (Fri, 28 Jul 2006) | 2 lines

just for fun, let's lock the whisper buffer for writing AND reading :-)

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r38452 | russell | 2006-07-28 18:14:49 -0400 (Fri, 28 Jul 2006) | 14 lines

Add the function attribute "pure" or "const" to various functions that perform
int to string or string to int operations.

"pure" essentially says that this function has no side effects aside from its
result, and the result depends on nothing else other than its arguments and
global variables.  "const" is a more strict form of "pure", where the function
also doesn't access any global variables.

>From the gcc manual: "Such a function can be subject to common subexpression 
elimination and loop optimization just as an arithmetic operator would be."
This also tells the compiler that it is safe to call the function fewer times
than the code says to, given the same arguments, since the result will always
be the same.

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r38454 | russell | 2006-07-28 18:50:54 -0400 (Fri, 28 Jul 2006) | 3 lines

add macros for the pure and const attributes to compiler.h, in case they ever
need to be handled differently for a specific compiler

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r38464 | russell | 2006-07-28 19:30:18 -0400 (Fri, 28 Jul 2006) | 4 lines

the pure attribute was not implemented before gcc 2.96.  Since we have this
version check here anwyay, define attribute_pure to be empty if it's an earlier
version.

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r38465 | kpfleming | 2006-07-28 19:36:06 -0400 (Fri, 28 Jul 2006) | 4 lines

add ExtenSpy variant of ChanSpy
implement whisper mode for ExtenSpy/ChanSpy


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r38466 | kpfleming | 2006-07-28 19:47:05 -0400 (Fri, 28 Jul 2006) | 2 lines

play a beep tone into the spied-on channel if we are about to whisper to them

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r38476 | kpfleming | 2006-07-28 20:19:18 -0400 (Fri, 28 Jul 2006) | 2 lines

update cleancount because the channel structure changed today

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r38489 | markster | 2006-07-29 18:02:37 -0400 (Sat, 29 Jul 2006) | 3 lines

Allow updates to match specific lines, allow specification of object or 
no when appending.

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r38502 | file | 2006-07-29 19:19:36 -0400 (Sat, 29 Jul 2006) | 9 lines

Blocked revisions 38501 via svnmerge

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r38501 | file | 2006-07-29 19:18:00 -0400 (Sat, 29 Jul 2006) | 2 lines

How many attempts does it take to make a SIP URI parser that works well? I'm up to 5 personally. On to the good stuff - parse the domain first, user second, and get rid of port & options/params last. (issue #7616 reported by andrew)

........

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r38521 | tilghman | 2006-07-30 11:02:34 -0400 (Sun, 30 Jul 2006) | 3 lines

Add details about flash and wait to doxygen about DTMF.
Reformat the lines to break at column 80.

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r38532 | russell | 2006-07-30 19:11:07 -0400 (Sun, 30 Jul 2006) | 2 lines

allow the gtk console to build and load

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r38533 | russell | 2006-07-30 19:16:23 -0400 (Sun, 30 Jul 2006) | 2 lines

suppress some compiler warnings ...

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r38548 | russell | 2006-07-31 00:10:43 -0400 (Mon, 31 Jul 2006) | 15 lines

Merged revisions 38546-38547 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38546 | russell | 2006-07-31 00:04:02 -0400 (Mon, 31 Jul 2006) | 2 lines

Make the frame counting done with TRACE_FRAMES defined thread-safe

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r38547 | russell | 2006-07-31 00:06:16 -0400 (Mon, 31 Jul 2006) | 2 lines

one more small tweak for thread-safety of TRACE_FRAMES

........

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r38573 | file | 2006-07-31 11:24:05 -0400 (Mon, 31 Jul 2006) | 2 lines

Poking at a structure when it may not even be allocated is not healthy. Essentially make sure an RTCP structure exists before trying to delete it's scheduled item. (issue #7514 reported by jmls fixed by AuPix)

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r38586 | file | 2006-07-31 13:11:51 -0400 (Mon, 31 Jul 2006) | 9 lines

Blocked revisions 38585 via svnmerge

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r38585 | file | 2006-07-31 13:09:10 -0400 (Mon, 31 Jul 2006) | 2 lines

Add missing code to bring transferee channel out of MOH/autoservice under certain circumstance (issue #7611 reported by guillecabeza with minor mods by myself)

........

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r38587 | russell | 2006-07-31 13:27:42 -0400 (Mon, 31 Jul 2006) | 2 lines

make the counting of ingress, outgress, and total frames thread-safe

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r38612 | kpfleming | 2006-07-31 17:31:44 -0400 (Mon, 31 Jul 2006) | 12 lines

Merged revisions 38611 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38611 | kpfleming | 2006-07-31 16:14:11 -0500 (Mon, 31 Jul 2006) | 4 lines

don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough)
don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others)
use ast_softhangup_nolock() since we already hold the channel's lock

........

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r38633 | tilghman | 2006-08-01 09:15:15 -0400 (Tue, 01 Aug 2006) | 2 lines

Set permissions completely open and modify by umask

................
r38652 | murf | 2006-08-01 15:03:32 -0400 (Tue, 01 Aug 2006) | 3 lines

This branch will resolve the bug 7635.


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r38655 | file | 2006-08-01 15:21:30 -0400 (Tue, 01 Aug 2006) | 10 lines

Merged revisions 38654 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38654 | file | 2006-08-01 15:20:05 -0400 (Tue, 01 Aug 2006) | 2 lines

Close the stream when file based MOH stop. This won't get rid of their position in the file but it will cause the translation path to be setup again. (issue #7634 reported by asimpson)

........

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r38687 | kpfleming | 2006-08-01 19:09:28 -0400 (Tue, 01 Aug 2006) | 10 lines

Merged revisions 38686 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38686 | kpfleming | 2006-08-01 18:07:06 -0500 (Tue, 01 Aug 2006) | 2 lines

ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)

........

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r38714 | mogorman | 2006-08-01 21:00:24 -0400 (Tue, 01 Aug 2006) | 3 lines

dtmf support. not everything else, trying to clear out those other bugs
but more to come i guess.

................
r38732 | kpfleming | 2006-08-02 15:32:38 -0400 (Wed, 02 Aug 2006) | 11 lines

Merged revisions 38731 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38731 | kpfleming | 2006-08-02 14:29:56 -0500 (Wed, 02 Aug 2006) | 3 lines

fix brain-damage I introduced when trying to fix the CANCEL/BYE sending mechanism for pending INVITES
accept unknown 1xx responses as 183 responses (as RFC3261 mandates we should do)

........

................
r38762 | tilghman | 2006-08-03 01:24:43 -0400 (Thu, 03 Aug 2006) | 10 lines

Merged revisions 38761 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38761 | tilghman | 2006-08-03 00:22:19 -0500 (Thu, 03 Aug 2006) | 2 lines

Bug 7648 - Checking wrong count for plurality on new messages for Dutch language

........

................
r38785 | tilghman | 2006-08-03 09:16:00 -0400 (Thu, 03 Aug 2006) | 2 lines

Update statement on FreeTDS with realtime, due to revisions in res_odbc

................
r38801 | crichter | 2006-08-03 12:38:00 -0400 (Thu, 03 Aug 2006) | 25 lines

* removed pp_l2_check (fixed L2 bug in mISDNuser)
* added blocking flag to stack object. A port can be blocked/unblocked from the
  cli
* added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later	
  we can add a manager event for that)
* added block_on_alarm option, to block the port whenever a ALARM occurs
* added need_busy flag to indicate if we've sended a CONTROL_BUSY already
* changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in 
  recent asterisk ast_log messages..
* fixed a few ETSI state violations, especially when finishing calls in 
  different seldom states
* changed debug levels a lot to make the log more readable in low debuglevels
* some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still)
* removed the PRECONNECTED state stuff
* added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this
  creates a CHANISUNAVAIL instead of a NOANSWER
* removed the addr pointer from "misdn show stacks" that's not needed anymore 
  and makes the output more unreadable
* added cause saving on RELEASE/RELEASE_COMPLETE
* set cause to 16 on prepare_bc
* removed stack getting from ph_control functions, we don't really need it 
  there
* added beroec api


................
r38823 | tilghman | 2006-08-03 15:05:06 -0400 (Thu, 03 Aug 2006) | 2 lines

Update documentation on realtime; add a workaround for lack of realtime hints by using func_odbc

................
r38826 | file | 2006-08-03 15:55:05 -0400 (Thu, 03 Aug 2006) | 10 lines

Merged revisions 38825 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38825 | file | 2006-08-03 15:54:02 -0400 (Thu, 03 Aug 2006) | 2 lines

Treat the file as invalid if we have no valid formats for it (issue #7643 reported by KNK)

........

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r38853 | file | 2006-08-04 14:40:47 -0400 (Fri, 04 Aug 2006) | 2 lines

Don't assume that a referred by URI will always exist (issue #7641 reported and fixed by AuPix)

................
r38871 | file | 2006-08-04 19:17:40 -0400 (Fri, 04 Aug 2006) | 2 lines

Minor optimization but use a linkedlists.h macro that pops the head off the list instead of getting the first entry and then removing it.

................
r38887 | file | 2006-08-04 20:13:44 -0400 (Fri, 04 Aug 2006) | 2 lines

It is better to hold a lock then not hold a lock when you are supposed to.

................
r38905 | russell | 2006-08-05 01:26:29 -0400 (Sat, 05 Aug 2006) | 23 lines

Merged revisions 38903-38904 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines

suppress a compiler warning about the usage of a potentially uninitialized variable

........
r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines

Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event.  This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before.  This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)

Also, fix a couple minor bugs in usecount handling.  chan_iax2 could have
increased the usecount but then returned an error.  The place where chan_sip
increased the usecount did not call ast_update_usecount()

........

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r38929 | russell | 2006-08-05 02:39:43 -0400 (Sat, 05 Aug 2006) | 11 lines

Merged revisions 38928 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38928 | russell | 2006-08-05 02:37:59 -0400 (Sat, 05 Aug 2006) | 3 lines

make sure the priv-callerintros directory exists before trying to create a file
there (issue #7659, patch by hads, with some modifications by me)

........

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r38951 | russell | 2006-08-05 03:22:27 -0400 (Sat, 05 Aug 2006) | 11 lines

Merged revisions 38950 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r38950 | russell | 2006-08-05 03:21:12 -0400 (Sat, 05 Aug 2006) | 3 lines

don't advertise that this function can set a SIP header when it can only
do reads

........

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r38973 | russell | 2006-08-05 04:10:20 -0400 (Sat, 05 Aug 2006) | 11 lines

Merged revisions 38972 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38972 | russell | 2006-08-05 04:08:48 -0400 (Sat, 05 Aug 2006) | 3 lines

remove duplicate queue log entry when the caller exits on a timeout
(issue #7616, ppyy)

........

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r38994 | russell | 2006-08-05 05:03:27 -0400 (Sat, 05 Aug 2006) | 14 lines

Merged revisions 38982 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r38982 | russell | 2006-08-05 05:01:37 -0400 (Sat, 05 Aug 2006) | 6 lines

Always generate a Newstate event in ast_setstate() instead of making it a
Newchannel event if the state was AST_STATE_DOWN.  The Newchannel event will
always be generated in ast_request(), so this just causes a duplicated
Newchannel event in some cases.  
(issue #7506, repoted by capouch, fixed by me)

........

................
r39014 | russell | 2006-08-05 07:25:36 -0400 (Sat, 05 Aug 2006) | 3 lines

suppress a compiler warning about the usage of a potentially uninitialized
variable

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r39032 | russell | 2006-08-05 16:58:47 -0400 (Sat, 05 Aug 2006) | 5 lines

- add DIALING_OFFHOOK and PRERING to ast_state2str
- don't include a newline character when building a response for an unknown
  state
(issue #7645, jonty)

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r39057 | file | 2006-08-06 20:17:12 -0400 (Sun, 06 Aug 2006) | 10 lines

Merged revisions 39056 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r39056 | file | 2006-08-06 20:15:51 -0400 (Sun, 06 Aug 2006) | 2 lines

Reset our stream and vstream pointers back to NULL so that any generator that uses them (file based MOH) will not try to close them again. (issue #7668 reported by jmls)

........

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r39058 | russell | 2006-08-06 20:26:11 -0400 (Sun, 06 Aug 2006) | 5 lines

minor cleanups
- use appropriate types in some assignments
- use ast_strlen_zero()
- don't manually free cid fields since ast_set_callerid() will handle it

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r39082 | russell | 2006-08-06 21:32:51 -0400 (Sun, 06 Aug 2006) | 15 lines

Merged revisions 39081 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r39081 | russell | 2006-08-06 21:28:29 -0400 (Sun, 06 Aug 2006) | 7 lines

Fix a crash reported to me by hads on IRC.  This crash would occur with the use
of the "distinctiveringaftercid" option.  Also, on this user's system, the crash
would only occur when built without optimizations.  This is because the bug is
that the code would write past the end of an array that was allocated on the
stack, and the structure of the stack is different with or without optimizations
enabled.

........

................
r39109 | russell | 2006-08-07 00:15:52 -0400 (Mon, 07 Aug 2006) | 9 lines

Merge my applicationmap_fixup branch to address the issues described in this
post to the asterisk-dev mailing list:
  http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html

This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on.  I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.

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r39131 | crichter | 2006-08-07 05:09:17 -0400 (Mon, 07 Aug 2006) | 1 line

changed naming of mISDN channels, so that hinting works proper
................
r39132 | oej | 2006-08-07 05:22:43 -0400 (Mon, 07 Aug 2006) | 3 lines

Issue #7651 - always send subscription-state (AuPix)
  Small change to the fix in the report.

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r39151 | murf | 2006-08-07 08:59:47 -0400 (Mon, 07 Aug 2006) | 1 line

The contents of branch teams/murf/bug_7598 are being folded in here.
................
r39171 | oej | 2006-08-07 12:20:23 -0400 (Mon, 07 Aug 2006) | 2 lines

Fix accidental removal of if() in commit 38905

................
r39182 | oej | 2006-08-07 13:01:40 -0400 (Mon, 07 Aug 2006) | 4 lines

Add placeholder for sla.conf sample in configs/. Please update with
correct information not to overload the switchboard with calls to sales.
 Thank you! ;-)

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r39203 | murf | 2006-08-07 14:12:51 -0400 (Mon, 07 Aug 2006) | 1 line

This commit folds in changes to both stringfields (some enhancements to the ...field_set() macro, to optimize setting strings to empty, resetting strings to shorter contents, etc.) and to chan_iax2.c, to use stringfields in the user, peer, and pvt structs. Has been running stably on iaxtel, but while iaxtel has a large registration volume, it doesn't seem to have a high call volume. So far, it seems to reduce heap usage by over half. YMMV\! Please report any IAX bugs that might involve stringfields\!
................
r39206 | kpfleming | 2006-08-07 14:47:33 -0400 (Mon, 07 Aug 2006) | 2 lines

remove some extraneous 'followme' in prompt names

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r39229 | mogorman | 2006-08-07 17:15:28 -0400 (Mon, 07 Aug 2006) | 3 lines

Many many code cleanup changes given to me by Oej
Thanks, sorry I didn't put this in forever ago.

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r39248 | mogorman | 2006-08-07 18:03:50 -0400 (Mon, 07 Aug 2006) | 4 lines

This patch allows for a user to send messages and 
monitor a jabber connection over manager.
patches from 7673 and 7666 with minor changes.

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r39272 | russell | 2006-08-08 02:32:04 -0400 (Tue, 08 Aug 2006) | 56 lines

Merge team/russell/ast_verbose_threadstorage

- instead of defining a free() wrapper in a bunch of files, define it as
  ast_free() in utils.h and remove the copies from all the files.

- centralize and abstract the code used for doing thread storage. The code
  lives in threadstorage.h, with one function being implemented in utils.c.
  This new API includes generic thread storage as well as special functions
  for handling thread local dynamic length string buffers.

- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API

- Modify manager_event() to use thread storage. Instead of using a buffer of
  4096 characters as the workspace for building the manager event, use a thread
  local dynamic string.  Now there is no length limitation on the length of the
  body of a manager event.

- Significantly simplify the handling of ast_verbose() ...
  - Instead of using a static char buffer and a lock to make sure only one
    thread can be using ast_verbose() at a time, use a thread local dynamic
    string as the workspace for preparing the verbose message. Instead of
    locking around the entire function, the only locking done now is when the
    message has been built and is being deliviered to the list of registered
    verbose message handlers.
  - This function was doing a strdup() on every message passed to it and
    keeping a queue of the last 200 messages in memory. This has been
    completely removed. The only place this was used was that if there were
    any messages in the verbose queue when a verbose handler was registered,
    all of the messages in the queue would be fed to it.  So, I just made sure
    that the console verbose handler and the network verbose handler (for
    remote asterisk consoles) were registered before any verbose messages.
    pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
    startup, but I didn't feel the performance hit of this message queue was
    worth saving the initial verbose output for these very rarely used modules.
  - I have removed the last three arguments to the verbose handlers, leaving
    only the string itself because they aren't needed anymore. For example,
    ast_verbose had some logic for telling the verbose handler to add
    a newline if the buffer was completely full. Now that the buffer can grow
    as needed, this doesn't matter anymore.
  - remove unused function, ast_verbose_dmesg() which was to dispatch the
    message queue
  - Convert the list of verbose handlers to use the linked list macros.

- add missing newline characters to a few ast_verbose() calls

- convert the list of log channels to use the linked list macros in logger.c

- fix close_logger() to close all of the files it opened for logging

- update ast_log() to use a thread local dynamic string for its workspace
  for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
  system) allocated on the stack.  The dynamic string in this case is limited
  to only growing to a maximum size of BUFSIZ.

................
r39274 | russell | 2006-08-08 02:36:47 -0400 (Tue, 08 Aug 2006) | 3 lines

change constant name from MAXSTRLEN to INITLEN since it is not the maximum
length, it is the initial length of the string when first allocated

................
r39295 | crichter | 2006-08-08 05:19:06 -0400 (Tue, 08 Aug 2006) | 14 lines

* added possibility to change echocancel settings during calls
* removed the holded element from the chan_list struct, we know this from the 
  state already
* added a few tweaks to make HOLD/RETRIEVE work again (TRANSFER does not work
  yet)
* added possibility to debug mISDN frames via syslog
* added misdn_lib_port_is_blocked function to check if a port is blocked

[... 25558 lines stripped ...]


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