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r106501 | russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines
Merge changes from team/russell/g722-sillyness ...
Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
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r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar 2008) | 16 lines
Merged revisions 106437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines
Quell an annoying message that is likely to print every single time that
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.
Thanks to joetester on IRC for pointing this out
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r106399 | juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines
trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense
given our removal of deadagi as an actual application.
(closes issue #12161)
Reported by: explidous
Patches:
res_agi_12161.patch uploaded by juggie (license 24)
Tested by: juggie
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r106250 | qwell | 2008-03-05 17:21:52 -0600 (Wed, 05 Mar 2008) | 1 line
Add a cmenuselect/cmenuconfig, to force curses.
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r106240 | russell | 2008-03-05 16:50:00 -0600 (Wed, 05 Mar 2008) | 5 lines
Add the nmenuselect makefile targets. This is a newt menuselect interface which
is much sexier than the curses one that was written by seanbright. :)
(issue #12139)
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r106236 | mmichelson | 2008-03-05 16:33:05 -0600 (Wed, 05 Mar 2008) | 11 lines
Adding the Atxfer manager command. With this, you may initiate
an attended transfer over AMI
(closes issue #10585)
Reported by: ornati
Patches:
atxfer-trunk-r90428.diff uploaded by ornati (license 210)
(with modifications from me)
Tested by: putnopvut
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r106179 | mvanbaak | 2008-03-05 15:15:57 -0600 (Wed, 05 Mar 2008) | 12 lines
Blocked revisions 106178 via svnmerge
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r106178 | mvanbaak | 2008-03-05 22:12:36 +0100 (Wed, 05 Mar 2008) | 5 lines
document var_metric so no bugreports will come in when it's actually a configuration issue.
(issue #12151)
Reported and patched by: caio1982
1.4 patch by me
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r105984 | russell | 2008-03-04 22:34:29 -0600 (Tue, 04 Mar 2008) | 3 lines
- simplify a few statements with ARRAY_LEN()
- constify the stregy int to string mappings array
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r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines
Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines
Fix a bug that I just noticed in the RTP code. The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use. The len field
represents the number of ms of audio that the frame contains. It would have
set the value to be twice what it should be.
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r105864 | mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5 lines
There are several places in manager.c where BUFSIZ is used for a buffer
which will contain nowhere near that amount of data. This makes these buffers
more reasonably sized.
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r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
(related to issue #11972)
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r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines
Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
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r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines
Merged revisions 106038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
(closes issue #11917)
Reported by: mavetju
Tested by: mavetju
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r105591 | russell | 2008-03-03 22:31:29 -0600 (Mon, 03 Mar 2008) | 4 lines
Backport a minor bug fix from trunk that I found while doing random code
cleanup. Properly break out of the loop when a context isn't found when
verify that includes are valid.
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