1565 Commits (352ca6584bcb1aa2742da08fbadc46f6415e643c)

Author	SHA1	Message	Date
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	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
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	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
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
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
Mark Michelson	4232cef812	Add format attribute to printf-style functions in astmm.h ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	00222ab483	ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109451 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
Steve Murphy	4e29ae8806	Merged revisions 109309 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines (closes issue #11903) Reported by: atis Many thanks to atis for spotting this problem and reporting it. The fix was to straighten out how items are placed on and removed from the file stack. Regressions as well as the provided test case helped to straighten out all code paths. valgrind was used to make sure all memory allocated was freed. Sorry for not solving this earlier. I got distracted. Added the ntest23 regression test, which is mainly a copy of ntest22, but with a few juicy errors thrown in, to replicate the kind of error that atis spotted. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109357 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
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
Russell Bryant	928475694a	Doxygenify dnsmgr.h ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108857 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
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
Mark Michelson	a96b50580c	Merged revisions 108227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when it is appropriate and when it is not appropriate to use it. I also removed the part of the debug message that mentions that this is probably a bug because there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g. when the scheduler callback manually reschedules with a new id instead of returning non-zero to tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro is probably being used in several places where it shouldn't be. Also removed the redundant line, file, and function information since that is provided by ast_log. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108238 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
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
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
Kevin P. Fleming	5875146526	Merged revisions 107461 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar 2008) | 2 lines stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	90786d21c9	Merged revisions 107405 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar 2008) | 2 lines fix small bug in IMAP toolkit testing ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	bdad3c9889	(closes issue #6019) ... Reported by: ssokol Patches: 20080304__bug6019.diff.txt uploaded by Corydon76 (license 14) Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	af7e1964f2	Merged revisions 107016 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	377e51c4d4	(closes issue #6002) ... Reported by: rizzo Tested by: murf Proposal of the changes to be made, and then an announcement of how they were accomplished: http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html and: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html Here is a recap, file by file, of what I have done: pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set. Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it is just as necessary to have the TABLE available. This is because the list/table in question might not be the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global position when things are ready. We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing "find" and "create", as all existing usages used both in tandem anyway. pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and then call merge_contexts_and_delete, which will merge (now) existing contexts and priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will lock down the contexts, swap the lists and tables, and unlock (real quick), and then destroy the old dialplan. chan_sip.c chan_iax.c chan_skinny.c All the channel drivers that would add regcontexts now use the ast_context_find_or_create now. chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered. apps/app_meetme.c apps/app_dial.c apps/app_queue.c All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead. include/asterisk/pbx.h ast_context_create() is removed. Find_or_create_ is the new method. ast_context_find_or_create() interface gets the hashtab added. ast_merge_contexts_and_delete() gets the local hashtab arg added. ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking. ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like fashion make public. include/asterisk/pval.h ast_compile_ael2() interface changed to include the local hashtab table ptr. main/features.c For the sake of the parking context, we use ast_context_find_or_create(). main/pbx.c I changed all the "tree" names to "table" instead. That's because the original implementation was based on binary trees. (had a free library). Then I moved to hashtabs. Now, the names move forward too. refcount field added to contexts, so you can keep track of how many modules wanted this context to exist. Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added some calls to ast_verb(3,...) for debug messages Lots of little mods to ast_context_remove_extension2, which is now excersized in ways it was not previously; one definite bug fixed. find_or_create was upgraded to handle both local lists/tables as well as the globals. context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables ast_merge_contexts_and_delete() was heavily modified. ast_add_extension2() was also upgraded to handle changes. the context_destroy() code was re-engineered to handle the new way of doing things, by exten/prio instead of by context. res/ael/pval.c res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the interface to ast_compile_ael2(), to include the hashtab ptr. This ended up involving several external apps. The main gotcha was I had to include lock.h and hashtab.h in several places. As a side note, I tested this stuff pretty thoroughly, I replicated the problems originally reported by Luigi, and made triply sure that reloads worked, and everything worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into trunk, that did not appear in my tests of bug6002. How's this for verbose commit messages? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	121bc8141f	Merged revisions 106704 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines Change a warning message to a debug message. This is happening quite frequently, and it is not worth spamming users with these messages unless we are pretty confident that it should never happen. As it stands today, it _will_ and _does_ happen and until that gets cleaned up a reasonable amount on the development side, let's not spam the logs of everyone else. (closes issue #12154) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	496adc6fc0	Merged revisions 106235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8a411ccf83	Create a centralized configuration option for silencethreshold ... (closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7a3f642207	Merged revisions 106015 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log. (closes issue #12140) Reported by: slavon Patches: sch2.patch uploaded by slavon (license 288) (Patch slightly modified by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a760a033e9	Merged revisions 105932 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	96e04792bd	add a destroy API call for a server instance ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	cc55483858	More public API name changes to use an appropriate ast_ prefix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	efb1e30a38	Rename public object server_instance to ast_tcptls_server_instance ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	4de0d8368f	Merged revisions 105674 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	8372a9bf08	3) In addition to merging the changes below, change trunk back to a regular ... LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	7d1891d5c3	Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. ... (closes issue #11520) Reported by: pliew Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7da06e6cb8	Merged revisions 105116 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines Fix a bug in the lock tracking code that was discovered by mmichelson. The issue is that if the lock history array was full, then the functions to mark a lock as acquired or not would adjust the stats for whatever lock is at the end of the array, which may not be itself. So, do a sanity check to make sure that we're updating lock info for the proper lock. (This explains the bizarre stats on lock #63 in BE-396, thanks Mark!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4aff24881b	Bring Voicetronix driver up to date with current drivers ... (closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0036cde5f3	Rename version.h to ast_version.h. Next, I will be re-adding version.h as an ... automatically generated file like it used to be. This still needs to be there for modules that have to check it to compile against multiple asterisk versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	2a7eac9940	Add an 'e' option to ResetCDR which re-enables a CDR that has been disabled. ... (closes issue #11170) Reported by: kratzers Patches: ResetCDR.1.diff uploaded by kratzers (license 307) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	3a8756c9b4	Merged revisions 104119 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f92a3e119e	Move Originate to a separate privilege and require the additional System privilege to call out to a subshell. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	edad328409	Disable epoll as it has caused more obscure issues then any of my previous code. I will continue to work on it in a separate branch to make it stable for a release and test it against the following issues. ... (closes issue #11253) Reported by: falves11 (closes issue #11657) Reported by: davevg (closes issue #11033) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ea8c695a1c	Largely refactor DSP tone detection routines. ... Separate fax detection from digit detected. Added CED (called) tone detection for fax (previously, only CNG (calling) was supported). Separate DTMF/MF code paths where appropriate. Allow detection of arbitary tones. (closes issue #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	808ae65263	Add some documentation fixups ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e54da94808	Add a non-invasive API for application level manipulation of T38 on a channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it. ... (closes issue #11873) Reported by: dimas Patches: v4-t38-api.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e6a260c747	Add an API call (ast_async_parseable_goto) which parses a goto string and does an async goto instead of an explicit goto. ... (closes issue #11753) Reported by: johan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	26755e3882	Context tracing for channels ... (closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago