https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 lines
Save 1 whopping byte of allocated memory!
This looks like it may have been a chicken/egg scenario..
You had to call a cleanup func, because everything was allocated.
Then since you had to call a cleanup func, you were forced to allocate - ie; strdup("").
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines
If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2 lines
ast_copy_string is not available when LOW_MEMORY is used and things are being built in the utils directory, so we need to resort to the old method of strncpy. (issue #8579 reported by mottano)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines
move extern declaration for this option to a header file where it belongs
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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done for exactly how much memory is needed. This was suggested by Luigi on
the asterisk-dev mailing list. Thanks!
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
before calling it.
This allows generators to set it to NULL when they have nothing to
do there.
Later, the three copies of the code that releases a generator
should be moved to a function.
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On passing remove presumably duplicate code to generate
the message for the manager_hooks:
in the previous version, the message was almost the same as the one sent
to regular sessions, with the exception of the empty line at the end, and
a few (presumably unintentional) differences e.g. timestamps,
debugging, and lowercase headers for "event" and "privilege".
now we reuse the same message as before.
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Jason Parker
Kevin P. Fleming
Joshua Colp
Steve Murphy
Olle Johansson
Luigi Rizzo
Russell Bryant