https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
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/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Russell Bryant
Luigi Rizzo
Steve Murphy
Matt O'Gorman