https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines
The SIGHUP handler was implemented to allow admins to send SIGHUP to a running
Asterisk process to reload the configuration. However, doing the actual reload
in the signal handler itself is a very bad thing to do, because the reload
process includes calling non-reentrant functions such as malloc/calloc/etc.
If Asterisk is running in the background, then the reload will happen
immediately. However, if running in console mode, the reload doesn't work
until something is typed at the console. That sort of defeats the purpose,
but I don't see an easy way to get around it at this point.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when the WaitEvent callback gets called, then no event can happen because the
session can't be locked by another thread. Also, the session needs to be
locked in the HTTP callback when it reads out the output string. This fixes
the deadlock reported in both 8711 and 8934.
Regarding issue 8711, there still may be an issue. If there is a second action
requested before the processing of the first action is finished, there could
still be some corruption of the output string buffer used to build the result.
(issue #8711, #8934)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Specifically indicate to the compiler that the "dropem" variable only
needs one but.
- Change formatting to conform to coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
would allow itself to be overfilled (per the max_jitterbuf parameter). Now
it rejects any data over and above that size, and complains about it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines
Fix an issue related to synchronization of recordings when using Monitor().
The bug is a miscalculation of the amount to seek the stream for writing to
disk when the number of samples coming in and out of a channel do not match up.
(issue #8298, #8887, report and patch by guillecabeza, patch files created and
testing done by whoiswes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when sending some sort of response, or calling one of the manager action
callbacks. This resolves an issue where people using the GUI would get random
crashes when they start clicking around a lot.
(issue #8711, reported and debugged by zandbelt)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
initialized to the list head *after* locking the list. Also, lock the actions
list in one place it is being accessed where it was not being done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
curses, termcap, or tinfo are further passed along to the editline configure
script. This fixes some cross-compilation environments.
(issue #8637, reported by ovi, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) | 4 lines
When ast_strip_quoted was called with a zero-length string, it would treat a
NULL as if it were the quoting character (and would thus return the string
in memory immediately following the passed-in string).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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("").
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
