https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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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.

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 - 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.


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to fill the buffer is half the buffer size.


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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)


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

Fix a comment that is no longer true.
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(closes issue #11734, reported and patched by jaroth)



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(Closes issue #11726, closes issue #11731)


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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.


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

Remove other remnants of pbx_kdeconsole

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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)

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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)

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version.h.


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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.

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


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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)


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

Strip terminal sequences from the verbose messages

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(related to issue #11706)


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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)

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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.



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(closes issue #11718)
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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.

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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.
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r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan 2008) | 3 lines

Part 2 of app_queue doxygen improvements. Some smaller functions this time


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we use.



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r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | 2 lines

Fix saying the parking space number to the caller doing the parking ...

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- support scrolling of message window;
- simplify the code for creating a message window,
  and try it using a second one in the top of
  the keypad (where we echo the dialed number).

The 'skin' that supports these two windows will be
committed separately.

 


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r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) | 2 lines

report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found

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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines

Set the caller id within the gtalk_alloc function.

As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.

Closes issue #11549.
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a number to dial in the 'message' area under the
keypad.

Now you can make calls using the keypad as a regular phone
(or the keyboard for chars not present on the keypad)



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