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${ noResults }
1576 Commits (ef8007f6300f7cfcd1dae74548396d81ef9a30c5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Mark Michelson
|
e409a129af |
Merged revisions 114063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines Fix a race condition that may happen between a sip hangup and a "core show channel" command. This patch adds locking to prevent the resulting crash. (closes issue #12155) Reported by: tsearle Patches: show_channels_crash2.patch uploaded by tsearle (license 373) Tested by: tsearle ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
115d5024a1 |
Merged revisions 114051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jason Parker
|
51c92a4644 |
Merged revisions 114035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines Only try to prefix language if we are not using an absolute path (suffix it otherwise). en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff uploaded by qwell (license 4) Tested by: kuj, qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Joshua Colp
|
4a21c5dd22 |
Fix spelling of existent in a few places.
(closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
28bd5d88c1 |
There was a subtle logical difference between 1.4 and trunk with regards to how timeouts
were handled. In 1.4, if the absolute timeout were reached on a call, no matter what the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T' extension or hangup otherwise. The rearrangement of this function in trunk made this check only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned 1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will behave as they did in 1.4 (closes issue #11550) Reported by: pj Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
f5a151e525 |
Move AST_FEATURE_FLAG_* and FEATURE_RETURN_* to features.h so that they can be used by modules.
(closes issue #12384) Reported by: fnordian Patches: features.patch uploaded by fnordian (license 110) (patch modified by me, to give FEATURE_RETURN_* an AST_ prefix) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
d3355ff2ed |
Merged revisions 113402 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113403 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
dc8fe3910d |
Merged revisions 113296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute. (closes issue #12296) Reported by: jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
be02a94138 |
Merged revisions 113065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines This fix prevents a deadlock that was experienced in chan_local. There was deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
c7d51a7fc1 |
Put my slinfactory changes back in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Dwayne M. Hubbard
|
5e6d84eb69 |
sleep long enough for the zaptel timer error message to display before exit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
b7b2e732f0 |
Merged revisions 112711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines Pass in the path to Zaptel for systems that install Zaptel headers in a separate location. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112712 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Dwayne M. Hubbard
|
6dafddbe39 |
satisfy buildbot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Dwayne M. Hubbard
|
593dcbe311 |
add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
|
0e6140c564 |
Use a 32k file buffer on recordings, which increases the efficiency of file recording.
(closes issue #11962) Reported by: garlew Patches: recording.patch uploaded by garlew (license 376) bug-11962.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112564 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
2580dfc6fb |
Merged revisions 112468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines Fix a race condition in the manager. It is possible that a new manager event could be appended during a brief time when the manager is not waiting for input. If an event comes during this period, we need to set an indicator that there is an event pending so that the manager doesn't attempt to wait forever for an event that already happened. (closes issue #12354) Reported by: bamby Patches: manager_race_condition.diff uploaded by bamby (license 430) (comments added by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Terry Wilson
|
1eb31edde2 |
Re-add HTTP post support by moving to res_http_post.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Steve Murphy
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da41d47a83 |
Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Steve Murphy
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2fb0bfba35 |
(closes issue #12298)
Reported by: falves11
Patches:
12298.patch1 uploaded by murf (license 17)
Tested by: murf
I have hopes that the changes made over the last few days will
finalize and solidify this code. While there are bound to be
small tweaks still needed, I feel that the job (at last) is
somewhat completed. Finally, I had a chance to comprehend how
the scoring of extension patterns was done in the previous
version, and I've come very close to using the exact same
criteria in the new pattern matching code. The left-right
sorting is now replicated in the trie structure itself, such
that the first match found will the 'best' match. Compared
the results against 1.4 for several extensions. Replicated
falves11's setup and it works. Used some devious patterns
provided by jsmith, supplemented with a few of my own.
Looks good.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
18 years ago |
|
Joshua Colp
|
0d7cfae6b6 |
Merged revisions 112209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112210 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
|
a5cdd849e5 |
This adds DNS SRV record support to DNS manager. If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
4dbacf6bbc |
Merged revisions 112138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines Initialize the __res_state structure used for dns purposes to all 0's prior to using it. This is due to valgrind's complaints on issue #12284 as well as an excerpt found in "Description" portion of the online man page found here: http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV (pertains to issue #12284 but does not necessarily close it) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
7dab892401 |
Merged revisions 112125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines Ensure that we do not exceed the hold's maximum size with a single frame. (closes issue #12047) Reported by: fabianoheringer Tested by: fabianoheringer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
|
f02c11d88b |
Yeah, simplify that logic a bit...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
|
aa720d402b |
Handle blank prefix= in http.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
|
16b2720cd4 |
Note a minor race condition that I noticed while reviewing Jeff's changes
to this code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
|
2848068017 |
Fix another little http problem. In making it match coding guidelines, a comparison was dropped
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Steve Murphy
|
3d4cb09ae8 |
comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
42358325a8 |
Merged revisions 111442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines For FreeBSD, at least, the ifa_addr element could be NULL. (closes issue #12300) Reported by: festr Patches: acl.c.patch uploaded by festr (license 443) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111443 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Steve Murphy
|
6928ccfa02 |
Merged revisions 111391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
8f2ae67a3e |
But we can change the API here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
0271088279 |
Merged revisions 111280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line Put this flag back so we don't change the API. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
f59c496a81 |
Merged revisions 111245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
|
4c2531989a |
Stupid strcasecmp function :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
e04025ead9 |
Simplify new macro, simplify configfile logic, now that list is sorted
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
e6fc9ae52c |
Add a linkedlist macro that maintains a sorted list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
dd2700d0b1 |
Only try to detect silence when we actually need to, instead of...always.
If this is wrong, I'd love to hear why. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
6412a96e43 |
Large cleanup of DSP code
Per comments from dimas: 1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before. 3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best. Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too. 4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used. 5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel. This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality. (closes issue #11968) Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas (license 88) v4-11968-zap.patch uploaded by dimas (license 88) Tested by: dimas, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
ef4eff9a9b |
Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
43d70915bb |
This ensures that the manager interface is not enabled by default. Prior to this
change, it was possible to start Asterisk with the manager interface enabled, then either comment out the enabled option or make manager.conf unopenable and the manager interface would still be enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
358ac2f76a |
Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Joshua Colp
|
30d85b3144 |
Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
6430ec3294 |
Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread. This really should not make a difference except in very rare cases. That case would be that all of the channels in autoservice are not generating any frames. In that case, this change reduces the potential amount of time that a thread waits in ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning of its loop. (closes issue #12266, reported by dimas) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
4e72f83d3e |
Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz. The issue here is that format modules give a "whennext" sample value, which is used to calculate when to set a timer for to retrieve the next frame. However, the zaptel timer operates on 8 kHz samples, so this must be taken into account. (another part of issue #12164, reported by milazzo and jsmith, patch by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
ff9befa36a |
Add missing unlock
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
bccebdd21f |
Remove astobj.h from some places where it wasn't needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
3c6cf5dcc5 |
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me. (issue #12164, reported by milazzo and jsmith, patches by me) res/res_musiconhold.c: - I moved a single line so that the sample queue update happened before ast_write(). The reason that this was a bug is that the G.722 frame originally says it has 320 samples in it (which is correct). However, when the frame is written to a channel that uses RTP, main/rtp.c modifies the frame to cut the number of samples in half before it sends it on the wire. This is to account for the stupid incorrect G.722 spec that makes it so we have to lie about the number of samples with RTP. I should probably go and re-work the RTP code so it doesn't modify the frame so that a bug like this won't happen in the future. However, this change to MOH is harmless. main/channel.c: - I made two fixes in regards to generator timing. Generators use samples for timing. However, this code assumed 8 kHz samples. In one case, it was a hard coded 160 samples, that is now written as the sample rate / 50. The other place was dealing with timing a generator based on frames coming from the other direction. However, that would have only worked if the sample rates for the formats in both directions were the same. The code now takes into account that the sample rates may differ, and scales the generator samples accordingly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
9e3603dac9 |
Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading. Early cleanup for issue #11968 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
8d4276578a |
Rename very poorly named function to reflect what it actually does. This was causing quite a bit of confusion for me...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |