asterisk

Commit Graph

Author	SHA1	Message	Date
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
Joshua Colp	3e439e9616	Merged revisions 110019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 \| file \| 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) \| 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e097cc7221	Add the ability to use a pattern match for a hint. (closes issue #7767) Reported by: Corydon76 Patches: 20070314__simple_hint_lookup.diff.txt uploaded by Corydon76 pbx-trunk-98436.diff uploaded by plack (license 365) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	14e1d8c6d8	Merged revisions 109908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109908 \| murf \| 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) \| 72 lines (closes issue #11442) Reported by: tzafrir Patches: 11442.patch uploaded by murf (license 17) Tested by: murf I didn't give tzafrir very much time to test this, but if he does still have remaining issues, he is welcome to re-open this bug, and we'll do what is called for. I reproduced the problem, and tested the fix, so I hope I am not jumping by just going ahead and committing the fix. The problem was with what file_save does with templates; firstly, it tended to print out multiple options: [my_category](!)(templateref) instead of [my_category](!,templateref) which is fixed by this patch. Nextly, the code to suppress output of duplicate declarations that would occur because the reader copies inherited declarations down the hierarchy, was not working. Thus: [master-template](!) mastervar = bar [template](!,master-template) tvar = value [cat](template) catvar = val would be rewritten as: ;! ;! Automatically generated configuration file ;! Filename: experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! [master-template](!) mastervar = bar [template](!,master-template) mastervar = bar tvar = value [cat](template) mastervar = bar tvar = value catvar = val This has been fixed. Since the config reader 'explodes' inherited vars into the category, users may, in certain circumstances, see output different from what they originally entered, but it should be both correct and equivalent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	75cb5032e6	actually implement HTTP request dispatching based on both URI and method; reduce duplication of data when generating responses using ast_http_error() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	4c6486782f	Fix some more breakage that I introduced when changing extension state callbacks to the list macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	84b133bd81	clean up code to conform to coding guidelines git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109909 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	89ad4ace67	Remove an unneeded variable. This compiled, but I missed the uninitialized warning because I always compile without optimizations turned on. Sorry! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b47eee2187	Convert handling of extension state callbacks to the list macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e1bd198bc0	Minor coding style changes, including adding handling for memory allocation failure git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f1d2a11aad	Minor change to use Asterisk macros git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c1cf92d304	Merged revisions 109838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109838 \| russell \| 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) \| 2 lines Tweak spacing in a recent change because I'm very picky. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	e191b51a08	start the process of changing HTTP request dispatching to do it based on both URI and method, so that POST support can move into a module; move http.c's private function prototypes into _private.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	b02bc230af	Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	760fc3403c	Make sure values are interpreted as character strings and not format strings. (AST-2008-004) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	10cdbe28a8	Merged revisions 109386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 \| file \| 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) \| 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	e727d15d34	Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory. This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros. If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	b246e010d5	Merged revisions 109226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109226 \| mmichelson \| 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) \| 12 lines Fix a logic flaw in the code that stores lock info which is displayed via the "core show locks" command. The idea behind this section of code was to remove the previous lock from the list if it was a trylock that had failed. Unfortunately, instead of checking the status of the previous lock, we were referencing the index immediately following the previous lock in the lock_info->locks array. The result of this problem, under the right circumstances, was that the lock which we currently in the process of attempting to acquire could "overwrite" the previous lock which was acquired. While this does not in any way affect typical operation, it could lead to misleading "core show locks" output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	0af58d3f5c	(closes issue #12238 ) Reported by: mvanbaak Tested by: murf, mvanbaak Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the includes, ignorepats, and switches. I added code to do this immediately after the context is created. This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully, this is pretty rare thing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	b311134430	Merged revisions 108961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108961 \| mvanbaak \| 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) \| 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	072eb8a913	Remove a double write lock of the contexts lock in ast_wrlock_contexts(). How did this ever work? (closes issue #12219) Reported by: ys Patches: pbx.c.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	963a2cec51	Make this compile git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	835df7d30f	Merged revisions 108583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108583 \| russell \| 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) \| 11 lines Fix another issue that was causing crashes in chanspy. This introduces a new datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	d236e3d1b1	Fixing a potential buffer overflow in the manager command ModuleCheck. Though this overflow is exploitable remotely, we are NOT issuing a security advisory for this since in order to exploit the overflow, the attacker would have to establish an authenticated manager session AND have the system privilege. By gaining this privilege, the attacker already has more powerful weapons at his disposal than overflowing a buffer with a malformed manager header, so the vulnerability in this case really lies with the authentication method that allowed the attacker to gain the system privilege in the first place. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c3c56990df	Make the default prefix empty, like it was in Asterisk 1.4. (closes issue #12198, reported by bkruse, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	8bbef5f996	Rename ast_tcptls_server_instance to session_instance, since this pertains to server and client usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	a3c7b08d19	Doxygenify slinfactory a bit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b38cb44acd	Merged revisions 108135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108135 \| russell \| 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) \| 40 lines (closes issue #12187, reported by atis, fixed by me after some brainstorming on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) \|\| Thread 2 (chanspy) --------------------------------------\|\|------------------------------------- ast_channel_free() \|\| - remove channel from channel list \|\| - lock/unlock the channel to ensure \|\| that no references retrieved from \|\| the channel list exist. \|\| --------------------------------------\|\|------------------------------------- \|\| channel_spy() - destroy some channel data \|\| - Lock chanspy datastore \|\| - Retrieve reference to channel \|\| - lock channel \|\| - Unlock chanspy datastore --------------------------------------\|\|------------------------------------- - destroy channel datastores \|\| - call chanspy datastore d'tor \|\| which NULL's out the ds' \|\| - Operate on the channel ... reference to the channel \|\| \|\| - free the channel \|\| \|\| \|\| - unlock the channel --------------------------------------\|\|------------------------------------- ============================================================================= ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	5fc569f5f5	Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 \| file \| 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) \| 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f7e28b12fe	Merged revisions 108031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108031 \| russell \| 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) \| 4 lines Destroy the channel lock after the channel datastores. (inspired by issue #12187) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	10609251f9	Revert several changes from revision 102525, as the changes were not compatible, and, in fact, introduced regressions. (Closes issue #12190) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	d02f74ebfe	An offhand comment from Russell made me realize that the configuration file caching would not work properly for users.conf and any other file read from more than one place. I needed to add the filename which requested the config file to get it to work properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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