[asterisk-commits] twilson: branch jpeeler/srtp r112557 - in /team/jpeeler/srtp: ./ apps/ build_...

[SVN commits to the Asterisk project]

Author: twilson
Date: Wed Apr  2 18:44:19 2008
New Revision: 112557

URL: http://svn.digium.com/view/asterisk?view=rev&rev=112557
Log:
Merged revisions 109227,109229,109282,109316,109357,109389-109390,109394,109396,109447,109451,109475,109545,109576,109621,109651,109680-109681,109683,109714,109762,109764,109775,109802,109833,109839,109841-109842,109883,109907,109909-109910,109912,109926,109942,109970,109974,110020,110023,110036,110084,110087,110132,110161,110164,110211,110237,110268,110270,110272,110303,110337,110339,110396,110444,110475,110499,110542,110578,110610,110615,110619,110621,110625,110629,110631,110636,110639,110689,110691,110726,110780,110831,110881,110911,110930,110963,111012-111013,111017,111021-111022,111025,111028,111036,111067,111083,111123,111127,111130,111132,111185,111213,111246,111285,111295,111360,111410,111443,111497,111500,111533,111565,111606,111659,111662,111721,111773-111774,111777,111811,111857,111908-111909,111961,111996,111998,112033,112035,112069,112071,112124,112126,112148,112155,112205,112207,112210,112234,112241,112252,112289,112321,112351,112357,112360,112394,112426,112431,
 112469,112520 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar 2008) | 20 lines

Merged revisions 109226 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.



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r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines

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.

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r109282 | seanbright | 2008-03-17 19:28:39 -0500 (Mon, 17 Mar 2008) | 1 line

Fix a typo
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r109316 | oej | 2008-03-18 02:23:45 -0500 (Tue, 18 Mar 2008) | 7 lines

Add manager peerstatus events when peer can't authenticate.

(closes issue #11959)
Reported by: mostyn
Patches: 
      peerstatus3.patch uploaded by mostyn (license 398)

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r109357 | murf | 2008-03-18 09:09:50 -0500 (Tue, 18 Mar 2008) | 25 lines

Merged revisions 109309 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines

(closes issue #11903)
Reported by: atis

Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.

Sorry for not solving this earlier. I got distracted.

Added the ntest23 regression test, which is mainly a copy of ntest22, 
but with a few juicy errors thrown in, to replicate the kind of 
error that atis spotted.



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r109389 | qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines

Do not return with a successful authentication if the From header ends up empty.
(AST-2008-003)

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r109390 | file | 2008-03-18 10:08:09 -0500 (Tue, 18 Mar 2008) | 11 lines

Merged revisions 109386 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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r109394 | qwell | 2008-03-18 10:10:45 -0500 (Tue, 18 Mar 2008) | 1 line

Block this here.  Already committed.
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r109396 | file | 2008-03-18 10:13:07 -0500 (Tue, 18 Mar 2008) | 3 lines

Make sure values are interpreted as character strings and not format strings.
(AST-2008-004)

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r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines

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.

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r109451 | kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 lines

ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly

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r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar 2008) | 2 lines

fix up various warnings found via the addition of format string checking... some of these were really, really bad code

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r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines

Add format attribute to printf-style functions in astmm.h


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r109576 | mmichelson | 2008-03-18 12:59:18 -0500 (Tue, 18 Mar 2008) | 14 lines

Merged revisions 109575 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines

Make sure an agent doesn't try to send dtmf to a NULL channel

closes issue #12242
Reported by Yourname


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r109621 | mmichelson | 2008-03-18 13:58:42 -0500 (Tue, 18 Mar 2008) | 9 lines

Add option 'randomperiodicannounce' to queues.conf. Setting this will
allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.

(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut


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r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | 15 lines

Merged revisions 109648 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | 7 lines

Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations.

(closes issue #12253)
Reported by: fossil
Patches:
      log2comp.patch uploaded by fossil (license 140)

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r109680 | tilghman | 2008-03-18 14:53:02 -0500 (Tue, 18 Mar 2008) | 2 lines

Comment debug

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r109681 | mmichelson | 2008-03-18 15:02:26 -0500 (Tue, 18 Mar 2008) | 7 lines

Since a sip request's data field is now a stringfield, we not only have to check
if the string is zero-length, but also if the data field is non-null.

(closes issue #12250)
Reported by: caio1982


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r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) | 4 lines

Set protocol version, port number correctly.
(closes issue #12211, closes issue #12209)
 Reported by: sylvain

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r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar 2008) | 20 lines

Merged revisions 109713 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines

This patch makes it so that all queue member status changes are handled through device state
code. This removes several problems people were seeing where their queue members would get into
an "unknown" state. Huge props go to atis on this one since he was the one who found the code
section that was causing the problem and proposed the solution. I just wrote what he suggested :)

(closes issue #12127)
Reported by: atis
Patches:
      12127v3.patch uploaded by putnopvut (license 60)
Tested by: atis, jvandal


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r109762 | kpfleming | 2008-03-18 17:32:26 -0500 (Tue, 18 Mar 2008) | 2 lines

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

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r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) | 11 lines

Merged revisions 109763 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) | 3 lines

Fix one place where the chanspy datastore isn't removed from a channel.
(issue #12243, reported by atis, patch by me)

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r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines

Change back to using ldap_initialize() and let the user specify a URL directly,
instead of trying to piece it together, badly.

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r109802 | mmichelson | 2008-03-18 18:32:58 -0500 (Tue, 18 Mar 2008) | 5 lines

Fix a typo which caused a double free in chan_zap. This was discovered
by Juggie while attempting to load chan_zap. Apparently this would happen
if an error were encountered while trying to process zapata.conf.


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r109833 | russell | 2008-03-18 22:51:06 -0500 (Tue, 18 Mar 2008) | 3 lines

Set req->data to NULL after free'ing to ensure that it never gets accidentally
double free'd.  (reported by dhubbard directly to me)

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r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) | 10 lines

Merged revisions 109838 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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r109841 | russell | 2008-03-18 23:09:55 -0500 (Tue, 18 Mar 2008) | 2 lines

Minor change to use Asterisk macros

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r109842 | russell | 2008-03-18 23:14:12 -0500 (Tue, 18 Mar 2008) | 2 lines

Minor coding style changes, including adding handling for memory allocation failure

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r109883 | russell | 2008-03-18 23:32:13 -0500 (Tue, 18 Mar 2008) | 2 lines

Convert handling of extension state callbacks to the list macros.

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r109907 | russell | 2008-03-19 10:22:13 -0500 (Wed, 19 Mar 2008) | 3 lines

Remove an unneeded variable.  This compiled, but I missed the uninitialized warning
because I always compile without optimizations turned on.  Sorry!

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r109909 | kpfleming | 2008-03-19 10:41:54 -0500 (Wed, 19 Mar 2008) | 2 lines

clean up code to conform to coding guidelines

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r109910 | russell | 2008-03-19 10:45:49 -0500 (Wed, 19 Mar 2008) | 2 lines

Fix some more breakage that I introduced when changing extension state callbacks to the list macros.

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r109912 | kpfleming | 2008-03-19 11:18:29 -0500 (Wed, 19 Mar 2008) | 2 lines

actually implement HTTP request dispatching based on both URI and method; reduce duplication of data when generating responses using ast_http_error()

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r109926 | kpfleming | 2008-03-19 11:21:36 -0500 (Wed, 19 Mar 2008) | 2 lines

ensure that res_phoneprov's HTTP handler tells the dispatcher what method it can handle

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r109942 | murf | 2008-03-19 11:24:51 -0500 (Wed, 19 Mar 2008) | 80 lines

Merged revisions 109908 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.



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r109970 | file | 2008-03-19 11:54:12 -0500 (Wed, 19 Mar 2008) | 7 lines

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)

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r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 109973 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | 5 lines

People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly...

Now we only show certain cflags when you run configure with --enable-dev-mode
(corresponding menuselect change to follow)

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r110020 | file | 2008-03-19 13:25:33 -0500 (Wed, 19 Mar 2008) | 14 lines

Merged revisions 110019 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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r110023 | russell | 2008-03-19 13:57:16 -0500 (Wed, 19 Mar 2008) | 2 lines

remove svnmerge-blocked property that is not supposed to be here

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r110036 | file | 2008-03-19 14:13:39 -0500 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110035 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky

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r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


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r110087 | jpeeler | 2008-03-19 16:05:24 -0500 (Wed, 19 Mar 2008) | 2 lines

This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.

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r110132 | qwell | 2008-03-19 16:56:15 -0500 (Wed, 19 Mar 2008) | 1 line

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...
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r110161 | qwell | 2008-03-19 17:25:34 -0500 (Wed, 19 Mar 2008) | 5 lines

Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue #11968

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r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 110163 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue #11990, reported by anakaoka, patched by me)

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r110211 | tilghman | 2008-03-19 22:14:59 -0500 (Wed, 19 Mar 2008) | 2 lines

Fix recent trunk breakage

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r110237 | tilghman | 2008-03-20 00:06:12 -0500 (Thu, 20 Mar 2008) | 5 lines

Upgrade the sounds version; add several directory enhancements:
	1) Number of digits to enter can now be configured
	2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue #7151)

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r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines

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.

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r110270 | russell | 2008-03-20 12:45:29 -0500 (Thu, 20 Mar 2008) | 2 lines

Remove astobj.h from some places where it wasn't needed

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r110272 | mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 lines

Add missing unlock


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r110303 | russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines

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)

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r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) | 22 lines

Merged revisions 110336 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

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r110339 | russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines

Use the correct buffer for g722tolin16_sample.  This shouldn't have caused any
problems, but Qwell noticed the typo here.

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r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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r110444 | tilghman | 2008-03-20 20:44:38 -0500 (Thu, 20 Mar 2008) | 2 lines

Add note of the added Directory options, from commit 110237 (closes issue #7151)

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r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | 15 lines

Merged revisions 110474 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines

Don't attempt to do optimizations of gsm on mips platforms either.

(closes issue #12270)
Reported by: zandbelt
Patches:
      026-gsm-mips.patch uploaded by zandbelt (license 33)

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r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines

Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues.

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r110542 | file | 2008-03-21 12:58:59 -0500 (Fri, 21 Mar 2008) | 2 lines

Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.

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r110578 | qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line

Update to 1.4.11 core sounds.
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r110610 | file | 2008-03-24 10:28:25 -0500 (Mon, 24 Mar 2008) | 6 lines

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue #12280)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)

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r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines

Merged revisions 110614 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

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r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines

Merged revisions 110618 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk 
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue #12272)
Reported by: qq12345


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r110621 | mmichelson | 2008-03-24 15:14:07 -0500 (Mon, 24 Mar 2008) | 11 lines

Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue #12279)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)


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r110625 | oej | 2008-03-25 05:54:07 -0500 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue #12278)
Reported by: rjain
Patches: 
      chan_sip.c.diff uploaded by rjain (license 226)

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r110629 | file | 2008-03-25 09:39:45 -0500 (Tue, 25 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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r110631 | file | 2008-03-25 10:18:41 -0500 (Tue, 25 Mar 2008) | 4 lines

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

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r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines

Merged revisions 110635 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


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r110639 | mmichelson | 2008-03-25 10:44:01 -0500 (Tue, 25 Mar 2008) | 3 lines

Oops here too. I need to stop coding for a while...


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r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines

Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293)
 Reported by: pprindeville
 Patches: 
       bugid-0012293.1.6.patch uploaded by pprindeville (license 347)

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r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines

Update sample configurations to make virtual hosting more obvious.
(closes issue #11969)
 Reported by: pprindeville
 Patches: 
       acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)

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r110726 | jpeeler | 2008-03-25 15:02:57 -0500 (Tue, 25 Mar 2008) | 2 lines

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.

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r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | 14 lines

Merged revisions 110779 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines

Make file access in cdr_custom similar to cdr_csv.

Fixes issue #12268.

Patch borrowed from r82344

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r110831 | mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 lines

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.


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r110881 | kpfleming | 2008-03-26 12:10:28 -0500 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 110880 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines

due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves

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r110911 | juggie | 2008-03-26 12:24:54 -0500 (Wed, 26 Mar 2008) | 8 lines

update documentation to reflect the changes in the way configure detects net-snmp.

(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie

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r110930 | juggie | 2008-03-26 12:28:49 -0500 (Wed, 26 Mar 2008) | 1 line

revert something dumb, because i was running svn diff in a subfolder not the root of trunk, before doing my commit and did not see it
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r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110962 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines

add note that the user will need to enable codec_ilbc to get it to build

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r111012 | tilghman | 2008-03-26 13:39:06 -0500 (Wed, 26 Mar 2008) | 3 lines

Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file.

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r111013 | tilghman | 2008-03-26 13:41:27 -0500 (Wed, 26 Mar 2008) | 2 lines

Oops, fix this, too

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r111017 | file | 2008-03-26 13:42:52 -0500 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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r111021 | file | 2008-03-26 14:05:42 -0500 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 111020 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines

If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be.
(closes issue #11995)
Reported by: fall

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r111022 | qwell | 2008-03-26 14:05:51 -0500 (Wed, 26 Mar 2008) | 23 lines

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

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r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 111024 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111019 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines

add a script to make getting the iLBC source code simple for end users

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r111028 | qwell | 2008-03-26 14:16:31 -0500 (Wed, 26 Mar 2008) | 4 lines

Only try to detect silence when we actually need to, instead of...always.

If this is wrong, I'd love to hear why.

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r111036 | tilghman | 2008-03-26 14:19:31 -0500 (Wed, 26 Mar 2008) | 2 lines

Add a linkedlist macro that maintains a sorted list

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r111067 | mmichelson | 2008-03-26 14:26:23 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111049 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines

Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.

(closes issue #10487)
Reported by: ewilhelmsen


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r111083 | file | 2008-03-26 14:29:26 -0500 (Wed, 26 Mar 2008) | 4 lines

Add expiry value to the sip show subscriptions CLI command.
(closes issue #12025)
Reported by: agx

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r111123 | mmichelson | 2008-03-26 14:39:23 -0500 (Wed, 26 Mar 2008) | 12 lines

Merged revisions 111121 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines

This code change is made just for clarification. It does exactly
the same thing as before. It just doesn't look as wrong.


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r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar 2008) | 18 lines

Merged revisions 111126 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 111125 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines

update UPGRADE notes to document usage of the script

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r111130 | file | 2008-03-26 14:56:40 -0500 (Wed, 26 Mar 2008) | 14 lines

Merged revisions 111129 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines

Update autosupport script.
(closes issue #12310)
Reported by: angler
Patches:
      autosupport.diff uploaded by angler (license 106)

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r111132 | tilghman | 2008-03-26 14:58:09 -0500 (Wed, 26 Mar 2008) | 2 lines

Simplify new macro, simplify configfile logic, now that list is sorted

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r111185 | tilghman | 2008-03-26 15:34:05 -0500 (Wed, 26 Mar 2008) | 2 lines

Oops, missed one

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r111213 | twilson | 2008-03-26 16:23:29 -0500 (Wed, 26 Mar 2008) | 2 lines

Stupid strcasecmp function :-)

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r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines

Merged revisions 111245 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines

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.

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r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | 9 lines

Merged revisions 111280 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line

Put this flag back so we don't change the API.
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r111295 | qwell | 2008-03-26 19:27:35 -0500 (Wed, 26 Mar 2008) | 1 line

But we can change the API here.
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r111360 | murf | 2008-03-26 23:47:12 -0500 (Wed, 26 Mar 2008) | 23 lines

Merged revisions 111341 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines


(closes issue #12302)
Reported by: pj
Tested by: murf

These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, 
and ever after that, till the end of the exten, we substitute any ${EXTEN} 
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). 
The reason for this, is that because switches are coded using 
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value, 
which blows the minds of users.


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r111410 | murf | 2008-03-27 08:29:41 -0500 (Thu, 27 Mar 2008) | 17 lines

Merged revisions 111391 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines

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.


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r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) | 14 lines

Merged revisions 111442 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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