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the smoother.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@70360 f38db490-d61c-443f-a65b-d21fe96a405b
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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...)
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disabled
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to frame.c to avoid build-killing compiler warnings.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49020 f38db490-d61c-443f-a65b-d21fe96a405b
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initializers (they default to zero anyway)
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allocations that were done for caching purposes, so they don't look like memory leaks
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48987 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines
Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs)
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consistent with "category verb arguments"
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original patch by andrew, updated by mithraen)
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Maximum values were incorrect, which is why this is being put in 1.4
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variable. It couldn't actually happen, though.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43264 f38db490-d61c-443f-a65b-d21fe96a405b
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option for setting our own packetization as apposed to just doing
what is asked.
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for uses in cases where you *know* that it will do no good. This patch was
inspired by file for use in some work of his on mixmonitor/chanspy.
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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