[asterisk-commits] murf: branch murf/bug11210 r98359 - in /team/murf/bug11210: ./ apps/ build_to...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jan 11 13:58:24 CST 2008 

Author: murf
Date: Fri Jan 11 13:58:23 2008
New Revision: 98359

URL: http://svn.digium.com/view/asterisk?view=rev&rev=98359
Log:
Merged revisions 97620,97623,97634,97641,97643-97644,97646,97650-97653,97655-97657,97698,97745,97758,97767,97769,97804-97805,97825-97826,97846,97848,97850,97890,97933,97974-97975,97977-97978,98024,98026-98027,98047,98081,98083,98124,98152,98193,98218,98220,98267-98270,98308,98316 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r97620 | qwell | 2008-01-09 13:13:14 -0700 (Wed, 09 Jan 2008) | 9 lines

Merged revisions 97618 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line

Fix some locking and return value funkiness.  We really shouldn't be unlocking this lock inside of a function, unless we locked it there too.
........

................
r97623 | qwell | 2008-01-09 13:30:54 -0700 (Wed, 09 Jan 2008) | 14 lines

Merged revisions 97622 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

(closes issue #11718)
........
r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines

Correctly display a message if a command could not be found.
Also fix a comment which may have led to this happening.

Issue 11718, reported by kshumard.

........

................
r97634 | twilson | 2008-01-09 14:37:26 -0700 (Wed, 09 Jan 2008) | 10 lines

Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution.  It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.


................
r97641 | russell | 2008-01-09 15:27:50 -0700 (Wed, 09 Jan 2008) | 11 lines

Merged revisions 97640 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | 3 lines

Make pbx_gtkconsole build ... but doesn't actually load on my system still
(related to issue #11706)

........

................
r97643 | russell | 2008-01-09 15:51:15 -0700 (Wed, 09 Jan 2008) | 3 lines

re-add check for gtk1, which is used for pbx_gtkconsole
(related to issue #11706)

................
r97644 | russell | 2008-01-09 15:53:12 -0700 (Wed, 09 Jan 2008) | 2 lines

re-gen configure

................
r97646 | russell | 2008-01-09 16:02:19 -0700 (Wed, 09 Jan 2008) | 10 lines

Merged revisions 97645 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | 2 lines

Strip terminal sequences from the verbose messages

........

................
r97650 | russell | 2008-01-09 17:11:02 -0700 (Wed, 09 Jan 2008) | 2 lines

Ensure that libg722.a gets rebuilt if one of the files changes

................
r97651 | tilghman | 2008-01-09 17:12:35 -0700 (Wed, 09 Jan 2008) | 11 lines

Several manager changes:
1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).

(Closes issue #10386)

................
r97652 | russell | 2008-01-09 17:17:02 -0700 (Wed, 09 Jan 2008) | 2 lines

Fix this so it doesn't force codec_g722 to get relinked every time

................
r97653 | twilson | 2008-01-09 17:22:50 -0700 (Wed, 09 Jan 2008) | 2 lines

Attempt at making lookup_iface work under FreeBSD.  Not yet tested, but it compiles under OS X.  And still works under linux.

................
r97655 | tilghman | 2008-01-09 17:39:34 -0700 (Wed, 09 Jan 2008) | 2 lines

oops, missed the case of a 0 permission (which should mean everybody is allowed, not nobody)

................
r97656 | murf | 2008-01-09 17:50:39 -0700 (Wed, 09 Jan 2008) | 1 line

The fixes in this commit are mainly to allow compiling of trunk with --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know.
................
r97657 | russell | 2008-01-09 17:54:01 -0700 (Wed, 09 Jan 2008) | 3 lines

These prototypes are not supposed to be in asterisk.h.  They are already in
version.h.

................
r97698 | file | 2008-01-10 08:12:08 -0700 (Thu, 10 Jan 2008) | 14 lines

Merged revisions 97697 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan 2008) | 6 lines

Don't try to copy the category from the group if no category exists.
(closes issue #11724)
Reported by: IgorG
Patches:
      group_count.v1.patch uploaded by IgorG (license 20)

........

................
r97745 | russell | 2008-01-10 09:15:43 -0700 (Thu, 10 Jan 2008) | 12 lines

Merged revisions 97734 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | 4 lines

Remove pbx_kdeconsole from the tree.  It hasn't worked in ages, and nobody has
complained.
(closes issue #11706, reported by caio1982)

........

................
r97758 | russell | 2008-01-10 09:22:10 -0700 (Thu, 10 Jan 2008) | 10 lines

Merged revisions 97753 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines

Remove other remnants of pbx_kdeconsole

........

................
r97767 | russell | 2008-01-10 09:27:03 -0700 (Thu, 10 Jan 2008) | 5 lines

Use AST_EXT_TOOL_CHECK() for the GTK check again.  I changed this to an inline
implementation to fix a small bug, but after a discussion with rizzo, I went to
change it back.  Also, it turns out that the implementation of the macro already
supported what was needed to fix the problem.

................
r97769 | russell | 2008-01-10 09:43:31 -0700 (Thu, 10 Jan 2008) | 2 lines

spaces to tabs

................
r97804 | russell | 2008-01-10 10:30:24 -0700 (Thu, 10 Jan 2008) | 2 lines

minor formatting changes

................
r97805 | tilghman | 2008-01-10 10:38:36 -0700 (Thu, 10 Jan 2008) | 3 lines

Fix problem with timestr going out of scope
(Closes issue #11726, closes issue #11731)

................
r97825 | twilson | 2008-01-10 12:03:04 -0700 (Thu, 10 Jan 2008) | 2 lines

Check pointers before freeing (was getting WARNINGS under MALLOC_DEBUG)

................
r97826 | twilson | 2008-01-10 12:07:36 -0700 (Thu, 10 Jan 2008) | 2 lines

heh, remove patch to generated file.

................
r97846 | mmichelson | 2008-01-10 13:05:43 -0700 (Thu, 10 Jan 2008) | 5 lines

Use the appropriate line ending for the X-Asterisk-VM-Message-Type header.

(closes issue #11734, reported and patched by jaroth)


................
r97848 | qwell | 2008-01-10 13:13:24 -0700 (Thu, 10 Jan 2008) | 9 lines

Merged revisions 97847 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line

Fix a comment that is no longer true.
........

................
r97850 | murf | 2008-01-10 13:45:05 -0700 (Thu, 10 Jan 2008) | 9 lines

Merged revisions 97849 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line

This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least.
........

................
r97890 | murf | 2008-01-10 14:46:56 -0700 (Thu, 10 Jan 2008) | 9 lines

Merged revisions 97889 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line

Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro.
........

................
r97933 | mmichelson | 2008-01-10 14:58:47 -0700 (Thu, 10 Jan 2008) | 14 lines

Merged revisions 97925 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines

Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen
to leave a message.

(closes issue #11735, reported and patched by jamessan)


........

................
r97974 | russell | 2008-01-10 16:10:00 -0700 (Thu, 10 Jan 2008) | 3 lines

Fix the buffer_samples value.  For signed linear, the number of samples needed
to fill the buffer is half the buffer size.

................
r97975 | russell | 2008-01-10 16:16:09 -0700 (Thu, 10 Jan 2008) | 7 lines

Fix various issues in codec_g722.
 - The most common fix being made here is to fix all of the places where the
   number of output samples and output bytes gets updated in the translator
   state structure.
 - Fix a number of other places where the number of samples provided as an
   initialization value to a struct was incorrect.

................
r97977 | russell | 2008-01-10 16:33:24 -0700 (Thu, 10 Jan 2008) | 11 lines

Merged revisions 97976 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines

Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.

........

................
r97978 | tilghman | 2008-01-10 16:40:13 -0700 (Thu, 10 Jan 2008) | 14 lines

Merged revisions 97973 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines

1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running.  Don't just ignore it.
(Closes issue #11698)

........

................
r98024 | russell | 2008-01-10 17:12:22 -0700 (Thu, 10 Jan 2008) | 3 lines

Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.

................
r98026 | russell | 2008-01-10 17:15:58 -0700 (Thu, 10 Jan 2008) | 10 lines

Blocked revisions 98025 via svnmerge

........
r98025 | russell | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines

Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.

........

................
r98027 | russell | 2008-01-10 17:38:23 -0700 (Thu, 10 Jan 2008) | 9 lines

Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you
to set the qualify frequency.

(closes issue #11597)
Reported by: wilder
Patches:
      qualifyfreq5.patch uploaded by wilder (license 362)
	   -- with some mods by me

................
r98047 | mmichelson | 2008-01-10 17:54:54 -0700 (Thu, 10 Jan 2008) | 7 lines

Fix "core show translation" to not output information for "unknown" codecs.
This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.

(closes issue #11722, reported and initially patched by caio1982, final patch by me)


................
r98081 | russell | 2008-01-10 20:37:19 -0700 (Thu, 10 Jan 2008) | 6 lines

 - Fix the last set of places where incorrect assumptions were made about the
   sample length with g722.  It is _2_ samples per byte, not 1.  This was all
   over the place, and I believed it, and it is what caused me to take so long
   to figure out what was broken.
 - Update copyright information on codec_g722.

................
r98083 | russell | 2008-01-10 20:40:58 -0700 (Thu, 10 Jan 2008) | 9 lines

Blocked revisions 98082 via svnmerge

........
r98082 | russell | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines

Fix samples vs. length calculations for g722

........

................
r98124 | kpfleming | 2008-01-11 05:51:21 -0700 (Fri, 11 Jan 2008) | 3 lines

Ascom phones send Flash events as SIP INFO using '!' as the 'digit'


................
r98152 | tilghman | 2008-01-11 08:12:33 -0700 (Fri, 11 Jan 2008) | 2 lines

Documentation updates

................
r98193 | tilghman | 2008-01-11 09:08:43 -0700 (Fri, 11 Jan 2008) | 10 lines

Merged revisions 98164 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines

Back out changes from revision 97077, since it wasn't perfect

........

................
r98218 | russell | 2008-01-11 10:17:54 -0700 (Fri, 11 Jan 2008) | 4 lines

At one point during working on this module, I had the lin/lin16 versions of the
framein callbacks different.  However, they are now the same again, so remove
the duplicate code and use the same functions for the lin/lin16 versions.

................
r98220 | file | 2008-01-11 10:27:58 -0700 (Fri, 11 Jan 2008) | 12 lines

Merged revisions 98219 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines

Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely

........

................
r98267 | russell | 2008-01-11 11:26:32 -0700 (Fri, 11 Jan 2008) | 18 lines

Blocked revisions 98265 via svnmerge

........
r98265 | russell | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines

Backport the ability to set the ToS bits on Linux when not running as root.
Normally, we would not backport features into 1.4, but, I was convinced by the
justification supplied by the supplier of this patch.  He pointed out that this
patch removes a requirement for running as root, thus reducing the potential
impacts of security issues.

(closes issue #11742)
Reported by: paravoid
Patches:
      libcap.diff uploaded by paravoid (license 200)

........

................
r98268 | tilghman | 2008-01-11 11:30:00 -0700 (Fri, 11 Jan 2008) | 2 lines

Commit Nick Gorham's suggestion for timestamp fix

................
r98269 | tilghman | 2008-01-11 11:35:30 -0700 (Fri, 11 Jan 2008) | 2 lines

Port Nick Gorham's timestamp patch to adaptive_odbc, too

................
r98270 | russell | 2008-01-11 11:48:07 -0700 (Fri, 11 Jan 2008) | 7 lines

Fix a bus error that happened when asterisk was built with optimizations on 
with platforms that explode on unaligned access.  I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on.  If it makes
sense to you, feel free to enlighten me.  :)

(closes issue #11725, patched by me)

................
r98308 | russell | 2008-01-11 12:05:24 -0700 (Fri, 11 Jan 2008) | 9 lines

Kevin noted that the thing that I _actually_ changed here was that I converted
a value from a double, to a float, back to a double.  Sure enough, when I changed
my interim variable back to a double, it still blows up.  Switching all of these
to a float fixes the problem.  This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.

(related to issue #11725)

................
r98316 | mmichelson | 2008-01-11 12:12:05 -0700 (Fri, 11 Jan 2008) | 13 lines

Merged revisions 98315 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines

Properly report the hangup cause as no answer when someone does not answer

(closes issue #10574, reported by boch, patched by moy)


........

................

Added:
    team/murf/bug11210/configs/phoneprov.conf.sample
      - copied unchanged from r98316, trunk/configs/phoneprov.conf.sample
    team/murf/bug11210/doc/tex/phoneprov.tex
      - copied unchanged from r98316, trunk/doc/tex/phoneprov.tex
    team/murf/bug11210/phoneprov/
      - copied from r98316, trunk/phoneprov/
    team/murf/bug11210/phoneprov/000000000000-directory.xml
      - copied unchanged from r98316, trunk/phoneprov/000000000000-directory.xml
    team/murf/bug11210/phoneprov/000000000000-phone.cfg
      - copied unchanged from r98316, trunk/phoneprov/000000000000-phone.cfg
    team/murf/bug11210/phoneprov/000000000000.cfg
      - copied unchanged from r98316, trunk/phoneprov/000000000000.cfg
    team/murf/bug11210/phoneprov/polycom.xml
      - copied unchanged from r98316, trunk/phoneprov/polycom.xml
    team/murf/bug11210/res/res_phoneprov.c
      - copied unchanged from r98316, trunk/res/res_phoneprov.c
Removed:
    team/murf/bug11210/pbx/kdeconsole_main.cc
    team/murf/bug11210/pbx/pbx_kdeconsole.cc
    team/murf/bug11210/pbx/pbx_kdeconsole.h
Modified:
    team/murf/bug11210/   (props changed)
    team/murf/bug11210/CHANGES
    team/murf/bug11210/Makefile
    team/murf/bug11210/apps/app_followme.c
    team/murf/bug11210/apps/app_stack.c
    team/murf/bug11210/apps/app_voicemail.c
    team/murf/bug11210/build_tools/menuselect-deps.in
    team/murf/bug11210/cdr/cdr_adaptive_odbc.c
    team/murf/bug11210/cdr/cdr_odbc.c
    team/murf/bug11210/channels/chan_iax2.c
    team/murf/bug11210/channels/chan_sip.c
    team/murf/bug11210/channels/console_video.c
    team/murf/bug11210/codecs/Makefile
    team/murf/bug11210/codecs/codec_g722.c
    team/murf/bug11210/codecs/codec_resample.c
    team/murf/bug11210/configs/manager.conf.sample
    team/murf/bug11210/configs/modules.conf.sample
    team/murf/bug11210/configs/sip.conf.sample
    team/murf/bug11210/configure
    team/murf/bug11210/configure.ac
    team/murf/bug11210/doc/manager_1_1.txt
    team/murf/bug11210/doc/tex/asterisk.tex
    team/murf/bug11210/formats/format_sln16.c
    team/murf/bug11210/funcs/func_groupcount.c
    team/murf/bug11210/funcs/func_strings.c
    team/murf/bug11210/include/asterisk/autoconfig.h.in
    team/murf/bug11210/include/asterisk/frame.h
    team/murf/bug11210/include/asterisk/localtime.h
    team/murf/bug11210/include/asterisk/manager.h
    team/murf/bug11210/main/Makefile
    team/murf/bug11210/main/acl.c
    team/murf/bug11210/main/ast_expr2.fl
    team/murf/bug11210/main/ast_expr2f.c
    team/murf/bug11210/main/channel.c
    team/murf/bug11210/main/cli.c
    team/murf/bug11210/main/db.c
    team/murf/bug11210/main/frame.c
    team/murf/bug11210/main/manager.c
    team/murf/bug11210/main/pbx.c
    team/murf/bug11210/main/stdtime/localtime.c
    team/murf/bug11210/main/translate.c
    team/murf/bug11210/makeopts.in
    team/murf/bug11210/pbx/pbx_gtkconsole.c
    team/murf/bug11210/pbx/pbx_realtime.c
    team/murf/bug11210/res/Makefile
    team/murf/bug11210/res/ael/ael.flex
    team/murf/bug11210/res/ael/ael_lex.c
    team/murf/bug11210/res/res_features.c
    team/murf/bug11210/utils/astman.c

Propchange: team/murf/bug11210/
------------------------------------------------------------------------------
    automerge = on

Propchange: team/murf/bug11210/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.

Propchange: team/murf/bug11210/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Propchange: team/murf/bug11210/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Jan 11 13:58:23 2008
@@ -1,1 +1,1 @@
-/trunk:1-97596
+/trunk:1-98316

Modified: team/murf/bug11210/CHANGES
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/CHANGES?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/CHANGES (original)
+++ team/murf/bug11210/CHANGES Fri Jan 11 13:58:23 2008
@@ -35,6 +35,10 @@
   * Added Masquerade manager event for when a masquerade happens between
      two channels.
   * Added "manager reload" command for the CLI
+  * Lots of commands that only provided information are now allowed under the
+     Reporting privilege, instead of only under Call or System.
+  * The IAX* commands now require either System or Reporting privilege, to
+     mirror the privileges of the SIP* commands.
 
 Dialplan functions
 ------------------
@@ -116,6 +120,8 @@
     considered advanced, so don't use them unless you have a problem.
   * Added a dial string option to be able to set the To: header in an INVITE to any
     SIP uri.
+  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
+     the qualify frequency.
 
 IAX2 changes
 ------------
@@ -436,3 +442,11 @@
   * Added a new codec translation module, codec_resample, which re-samples
      signed linear audio between 8 kHz and 16 kHz to help support wideband
      codecs.
+  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
+     based on configuration templates that use Asterisk dialplan function and
+     variable substitution.  It should be possible to create phone profiles and
+     templates that work for the majority of phones provisioned over http. It
+     is currently only intended to provision a single user account per phone.
+     An example profile and set of templates for Polycom phones is provided.
+     NOTE: Polycom firmware is not included, but should be placed in
+     AST_DATA_DIR/phoneprov/configs to match up with the included templates. 

Modified: team/murf/bug11210/Makefile
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/Makefile?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/Makefile (original)
+++ team/murf/bug11210/Makefile Fri Jan 11 13:58:23 2008
@@ -451,6 +451,10 @@
 # Should static HTTP be installed during make samples or even with its own target ala
 # webvoicemail?  There are portions here that *could* be customized but might also be
 # improved a lot.  I'll put it here for now.
+	mkdir -p $(DESTDIR)$(ASTDATADIR)/phoneprov
+	for x in phoneprov/*; do \
+		$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/phoneprov ; \
+	done
 	mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http
 	for x in static-http/*; do \
 		$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/static-http ; \

Modified: team/murf/bug11210/apps/app_followme.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/apps/app_followme.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/apps/app_followme.c (original)
+++ team/murf/bug11210/apps/app_followme.c Fri Jan 11 13:58:23 2008
@@ -1012,7 +1012,6 @@
 		pbx_builtin_setvar_helper(caller, "ANSWEREDTIME", toast);
 		if (outbound)
 			ast_hangup(outbound);
-		res = 1;
 	}
 
 	outrun:

Modified: team/murf/bug11210/apps/app_stack.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/apps/app_stack.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/apps/app_stack.c (original)
+++ team/murf/bug11210/apps/app_stack.c Fri Jan 11 13:58:23 2008
@@ -100,7 +100,7 @@
 		} else
 			pbx_builtin_setvar_helper(chan, var, value);
 
-		manager_event(EVENT_FLAG_CALL, "VarSet", 
+		manager_event(EVENT_FLAG_DIALPLAN, "VarSet", 
 			"Channel: %s\r\n"
 			"Variable: LOCAL(%s)\r\n"
 			"Value: %s\r\n"

Modified: team/murf/bug11210/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/apps/app_voicemail.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/apps/app_voicemail.c (original)
+++ team/murf/bug11210/apps/app_voicemail.c Fri Jan 11 13:58:23 2008
@@ -2020,7 +2020,7 @@
 		fprintf(p, "X-Asterisk-VM-Duration: %d" ENDL, duration);
 		if (!ast_strlen_zero(category)) 
 			fprintf(p, "X-Asterisk-VM-Category: %s" ENDL, category);
-		fprintf(p, "X-Asterisk-VM-Message-Type: %s\n", msgnum > -1 ? "Message" : greeting_attachment);
+		fprintf(p, "X-Asterisk-VM-Message-Type: %s" ENDL, msgnum > -1 ? "Message" : greeting_attachment);
 		fprintf(p, "X-Asterisk-VM-Orig-date: %s" ENDL, date);
 		fprintf(p, "X-Asterisk-VM-Orig-time: %ld" ENDL, (long)time(NULL));
 	}
@@ -4300,8 +4300,8 @@
 		/* start optimistic */
 		valid_extensions = 1;
 		while (s) {
-			/* Don't forward to ourselves.  find_user is going to malloc since we have a NULL as first argument */
-			if (strcmp(s,sender->mailbox) && (receiver = find_user(NULL, context, s))) {
+			/* Don't forward to ourselves but allow leaving a message for ourselves (flag == 1).  find_user is going to malloc since we have a NULL as first argument */
+			if ((flag == 1 || strcmp(s,sender->mailbox)) && (receiver = find_user(NULL, context, s))) {
 				AST_LIST_INSERT_HEAD(&extensions, receiver, list);
 				found++;
 			} else {
@@ -8745,7 +8745,7 @@
 	res |= ast_register_application(app3, vm_box_exists, synopsis_vm_box_exists, descrip_vm_box_exists);
 	res |= ast_register_application(app4, vmauthenticate, synopsis_vmauthenticate, descrip_vmauthenticate);
 	res |= ast_custom_function_register(&mailbox_exists_acf);
-	res |= ast_manager_register("VoicemailUsersList", EVENT_FLAG_CALL, manager_list_voicemail_users, "List All Voicemail User Information");
+	res |= ast_manager_register("VoicemailUsersList", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, manager_list_voicemail_users, "List All Voicemail User Information");
 	if (res)
 		return res;
 

Modified: team/murf/bug11210/build_tools/menuselect-deps.in
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/build_tools/menuselect-deps.in?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/build_tools/menuselect-deps.in (original)
+++ team/murf/bug11210/build_tools/menuselect-deps.in Fri Jan 11 13:58:23 2008
@@ -24,7 +24,6 @@
 PORTAUDIO=@PBX_PORTAUDIO@
 PRI=@PBX_PRI@
 SS7=@PBX_SS7@
-QT=@PBX_QT@
 RADIUS=@PBX_RADIUS@
 SPEEX=@PBX_SPEEX@
 SQLITE=@PBX_SQLITE@

Modified: team/murf/bug11210/cdr/cdr_adaptive_odbc.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/cdr/cdr_adaptive_odbc.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/cdr/cdr_adaptive_odbc.c (original)
+++ team/murf/bug11210/cdr/cdr_adaptive_odbc.c Fri Jan 11 13:58:23 2008
@@ -404,8 +404,8 @@
 							year += 2000;
 
 						lensql += snprintf(sql + lensql, sizesql - lensql, "%s,", entry->name);
-						LENGTHEN_BUF2(10);
-						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "'%04d-%02d-%02d',", year, month, day);
+						LENGTHEN_BUF2(17);
+						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "{ d '%04d-%02d-%02d' },", year, month, day);
 					}
 					break;
 				case SQL_TYPE_TIME:
@@ -419,8 +419,8 @@
 						}
 
 						lensql += snprintf(sql + lensql, sizesql - lensql, "%s,", entry->name);
-						LENGTHEN_BUF2(8);
-						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "'%02d:%02d:%02d',", hour, minute, second);
+						LENGTHEN_BUF2(15);
+						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "{ t '%02d:%02d:%02d' },", hour, minute, second);
 					}
 					break;
 				case SQL_TYPE_TIMESTAMP:
@@ -445,8 +445,8 @@
 							year += 2000;
 
 						lensql += snprintf(sql + lensql, sizesql - lensql, "%s,", entry->name);
-						LENGTHEN_BUF2(19);
-						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "'%04d-%02d-%02d %02d:%02d:%02d',", year, month, day, hour, minute, second);
+						LENGTHEN_BUF2(26);
+						lensql2 += snprintf(sql2 + lensql2, sizesql2 - lensql2, "{ ts '%04d-%02d-%02d %02d:%02d:%02d' },", year, month, day, hour, minute, second);
 					}
 					break;
 				case SQL_INTEGER:

Modified: team/murf/bug11210/cdr/cdr_odbc.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/cdr/cdr_odbc.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/cdr/cdr_odbc.c (original)
+++ team/murf/bug11210/cdr/cdr_odbc.c Fri Jan 11 13:58:23 2008
@@ -68,18 +68,18 @@
 	SQLHSTMT stmt;
 
 	ast_localtime(&cdr->start, &tm, ast_test_flag(&config, CONFIG_USEGMTIME) ? "GMT" : NULL);
-
 	ast_strftime(timestr, sizeof(timestr), DATE_FORMAT, &tm);
+
 	if (ast_test_flag(&config, CONFIG_LOGUNIQUEID)) {
 		snprintf(sqlcmd,sizeof(sqlcmd),"INSERT INTO %s "
 		"(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,"
 		"lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) "
-		"VALUES (?,?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)", table);
+		"VALUES ({ts '%s'},?,?,?,?,?,?,?,?,?,?,?,?,?,?,?)", table, timestr);
 	} else {
 		snprintf(sqlcmd,sizeof(sqlcmd),"INSERT INTO %s "
 		"(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,"
 		"duration,billsec,disposition,amaflags,accountcode) "
-		"VALUES (?,?,?,?,?,?,?,?,?,?,?,?,?,?)", table);
+		"VALUES ({ts '%s'},?,?,?,?,?,?,?,?,?,?,?,?,?)", table, timestr);
 	}
 
 	ODBC_res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
@@ -98,27 +98,26 @@
 		return NULL;
 	}
 
-	SQLBindParameter(stmt, 1, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(timestr), 0, &timestr, 0, NULL);
-	SQLBindParameter(stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, 0, NULL);
-	SQLBindParameter(stmt, 3, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->src), 0, cdr->src, 0, NULL);
-	SQLBindParameter(stmt, 4, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dst), 0, cdr->dst, 0, NULL);
-	SQLBindParameter(stmt, 5, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dcontext), 0, cdr->dcontext, 0, NULL);
-	SQLBindParameter(stmt, 6, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->channel), 0, cdr->channel, 0, NULL);
-	SQLBindParameter(stmt, 7, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dstchannel), 0, cdr->dstchannel, 0, NULL);
-	SQLBindParameter(stmt, 8, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->lastapp), 0, cdr->lastapp, 0, NULL);
-	SQLBindParameter(stmt, 9, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->lastdata), 0, cdr->lastdata, 0, NULL);
-	SQLBindParameter(stmt, 10, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->duration, 0, NULL);
-	SQLBindParameter(stmt, 11, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->billsec, 0, NULL);
+	SQLBindParameter(stmt, 1, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, 0, NULL);
+	SQLBindParameter(stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->src), 0, cdr->src, 0, NULL);
+	SQLBindParameter(stmt, 3, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dst), 0, cdr->dst, 0, NULL);
+	SQLBindParameter(stmt, 4, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dcontext), 0, cdr->dcontext, 0, NULL);
+	SQLBindParameter(stmt, 5, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->channel), 0, cdr->channel, 0, NULL);
+	SQLBindParameter(stmt, 6, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->dstchannel), 0, cdr->dstchannel, 0, NULL);
+	SQLBindParameter(stmt, 7, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->lastapp), 0, cdr->lastapp, 0, NULL);
+	SQLBindParameter(stmt, 8, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->lastdata), 0, cdr->lastdata, 0, NULL);
+	SQLBindParameter(stmt, 9, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->duration, 0, NULL);
+	SQLBindParameter(stmt, 10, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->billsec, 0, NULL);
 	if (ast_test_flag(&config, CONFIG_DISPOSITIONSTRING))
-		SQLBindParameter(stmt, 12, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(ast_cdr_disp2str(cdr->disposition)) + 1, 0, ast_cdr_disp2str(cdr->disposition), 0, NULL);
+		SQLBindParameter(stmt, 11, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(ast_cdr_disp2str(cdr->disposition)) + 1, 0, ast_cdr_disp2str(cdr->disposition), 0, NULL);
 	else
-		SQLBindParameter(stmt, 12, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->disposition, 0, NULL);
-	SQLBindParameter(stmt, 13, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->amaflags, 0, NULL);
-	SQLBindParameter(stmt, 14, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->accountcode), 0, cdr->accountcode, 0, NULL);
+		SQLBindParameter(stmt, 11, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->disposition, 0, NULL);
+	SQLBindParameter(stmt, 12, SQL_PARAM_INPUT, SQL_C_SLONG, SQL_INTEGER, 0, 0, &cdr->amaflags, 0, NULL);
+	SQLBindParameter(stmt, 13, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->accountcode), 0, cdr->accountcode, 0, NULL);
 
 	if (ast_test_flag(&config, CONFIG_LOGUNIQUEID)) {
-		SQLBindParameter(stmt, 15, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->uniqueid), 0, cdr->uniqueid, 0, NULL);
-		SQLBindParameter(stmt, 16, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->userfield), 0, cdr->userfield, 0, NULL);
+		SQLBindParameter(stmt, 14, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->uniqueid), 0, cdr->uniqueid, 0, NULL);
+		SQLBindParameter(stmt, 15, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->userfield), 0, cdr->userfield, 0, NULL);
 	}
 
 	return stmt;

Modified: team/murf/bug11210/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/channels/chan_iax2.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/channels/chan_iax2.c (original)
+++ team/murf/bug11210/channels/chan_iax2.c Fri Jan 11 13:58:23 2008
@@ -11670,9 +11670,9 @@
 
 	ast_register_application(papp, iax2_prov_app, psyn, pdescrip);
 	
-	ast_manager_register( "IAXpeers", 0, manager_iax2_show_peers, "List IAX Peers" );
-	ast_manager_register( "IAXpeerlist", 0, manager_iax2_show_peer_list, "List IAX Peers" );
-	ast_manager_register( "IAXnetstats", 0, manager_iax2_show_netstats, "Show IAX Netstats" );
+	ast_manager_register( "IAXpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_iax2_show_peers, "List IAX Peers" );
+	ast_manager_register( "IAXpeerlist", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_iax2_show_peer_list, "List IAX Peers" );
+	ast_manager_register( "IAXnetstats", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_iax2_show_netstats, "Show IAX Netstats" );
 
 	if(set_config(config, 0) == -1)
 		return AST_MODULE_LOAD_DECLINE;

Modified: team/murf/bug11210/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug11210/channels/chan_sip.c?view=diff&rev=98359&r1=98358&r2=98359
==============================================================================
--- team/murf/bug11210/channels/chan_sip.c (original)
+++ team/murf/bug11210/channels/chan_sip.c Fri Jan 11 13:58:23 2008
@@ -139,7 +139,6 @@
 #include "asterisk/translate.h"
 #include "asterisk/version.h"
 #include "asterisk/event.h"
-#include "asterisk/astobj2.h"
 
 /* remove when someone fixes this */
 const char *ast_get_version(void);
@@ -187,7 +186,7 @@
 #define CALLERID_UNKNOWN        "Unknown"
 
 #define DEFAULT_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
-#define DEFAULT_FREQ_OK              60 * 1000        /*!< Qualification: How often to check for the host to be up */
+#define DEFAULT_QUALIFYFREQ          60 * 1000        /*!< Qualification: How often to check for the host to be up */
 #define DEFAULT_FREQ_NOTOK           10 * 1000        /*!< Qualification: How often to check, if the host is down... */
 
 #define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
@@ -654,8 +653,9 @@
 static int global_autoframing;          /*!< Turn autoframing on or off. */
 static enum transfermodes global_allowtransfer;	/*!< SIP Refer restriction scheme */
 static struct sip_proxy global_outboundproxy;	/*!< Outbound proxy */
-
 static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
+static int global_qualifyfreq; /*!< Qualify frequency */
+
 
 /*! \brief Codecs that we support by default: */
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
@@ -1164,7 +1164,7 @@
 	struct ast_rtp *rtp;			/*!< RTP Session */
 	struct ast_rtp *vrtp;			/*!< Video RTP session */
 	struct ast_rtp *trtp;			/*!< Text RTP session */
-	struct ao2_container *packets;		/*!< Packets scheduled for re-transmission */
+	struct sip_pkt *packets;		/*!< Packets scheduled for re-transmission */
 	struct sip_history_head *history;	/*!< History of this SIP dialog */
 	size_t history_entries;			/*!< Number of entires in the history */
 	struct ast_variable *chanvars;		/*!< Channel variables to set for inbound call */
@@ -1260,6 +1260,7 @@
  * require retransmissions.
  */
 struct sip_pkt {
+	struct sip_pkt *next;			/*!< Next packet in linked list */
 	int retrans;				/*!< Retransmission number */
 	int method;				/*!< SIP method for this packet */
 	int seqno;				/*!< Sequence number */
@@ -1387,6 +1388,7 @@
 	int pokeexpire;			/*!<  When to expire poke (qualify= checking) */
 	int lastms;			/*!<  How long last response took (in ms), or -1 for no response */
 	int maxms;			/*!<  Max ms we will accept for the host to be up, 0 to not monitor */
+	int qualifyfreq;		/*!<  Qualification: How often to check for the host to be up */
 	struct timeval ps;		/*!<  Time for sending SIP OPTION in sip_pke_peer() */
 	struct sockaddr_in defaddr;	/*!<  Default IP address, used until registration */
 	struct ast_ha *ha;		/*!<  Access control list */
@@ -2482,15 +2484,12 @@
 /*! \brief Retransmit SIP message if no answer (Called from scheduler) */
 static int retrans_pkt(const void *data)
 {
-	struct sip_pkt *pkt = (struct sip_pkt *)data, *prev;
+	struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
 	int reschedule = DEFAULT_RETRANS;
 	int xmitres = 0;
 	
-	ao2_ref(pkt, 1); /* Make sure this cannot go away while we're using it */
-
 	/* Lock channel PVT */
-	if (pkt->owner)
-		sip_pvt_lock(pkt->owner);
+	sip_pvt_lock(pkt->owner);
 
 	if (pkt->retrans < MAX_RETRANS) {
 		pkt->retrans++;
@@ -2517,7 +2516,7 @@
  			ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
  		} 
 
-		if (pkt->owner && sip_debug_test_pvt(pkt->owner)) {
+		if (sip_debug_test_pvt(pkt->owner)) {
 			const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
 			ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",

[... 5596 lines stripped ...]

More information about the asterisk-commits mailing list