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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98081 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98047 f38db490-d61c-443f-a65b-d21fe96a405b
to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
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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.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98026 f38db490-d61c-443f-a65b-d21fe96a405b
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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- 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97975 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97850 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97767 f38db490-d61c-443f-a65b-d21fe96a405b
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
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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- 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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97488 f38db490-d61c-443f-a65b-d21fe96a405b
