78 Commits (076b29dd5b51e7ec68c141d9e6c5ec8a2c63be66)

Author	SHA1	Message	Date
Andrew Latham	6c20cf2d8a	Doxygen Updates - Title update ... Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	90f8c90b10	dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END ... Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374481 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374485 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	36405ae659	dsp.c fix incorrect DTMF Digit_Duration. ... it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2145/ ........ Merged revisions 374475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374476 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374477 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	4af961a03a	dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values ... Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	368b4c6166	dsp.c: remove more whitespace mentioned in review2107 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	d3a23be26e	dsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup ... janitor cleanup. No functional change. 1). ast_dsp_call_progress: use 'short samp' instead of s[x] inside loop. apply same casting as other _init, dsp->energy = (int32_t) samp * (int32_t) samp 2). ast_dtmf_detect_init: move repeated setting of s->energy to outside of loop. do goertzel_init loop first before setting s->lasthit and s->current_hit, consistant with ast_dsp_digitreset() 3). ast_mf_detect_init: do goertzel_init loop first before setting s->hits[] and s->current_hit, consistant with ast_dsp_digitreset() 4). Don't chain init different variables, as the type may change Review https://reviewboard.asterisk.org/r/2107/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	ed442248e5	dsp.c: remove whitespace mentioned in review2107 ... Related https://reviewboard.asterisk.org/r/2107/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	d39b417074	dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, should be MF_GSIZE ... Remove unused goertzel_state_t member 'samples'. Related https://reviewboard.asterisk.org/r/2097/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	7af245a403	dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms ... Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary; 1. reseting of hits=0, when no signal, only need to set it once. 2. incrementing of hits, when the hit is the same as the current hit. 3. setting of lasthit, when it's the same as before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3 spelling mistakes (closes issue ASTERISK-19610) alecdavis (license 585) Reported by: Jean-Philippe Lord Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2085/ ........ Merged revisions 372239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372240 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372241 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	2f9f9ecffb	dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect ... use a temporary short int when repeatedly used to call goertzel_sample. alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2093/ ........ Merged revisions 372212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372213 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372214 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1295d551f9	mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE ... ........ Merged revisions 371662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371663 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371664 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() ... This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	166b4e2b30	Multiple revisions 369001-369002 ... ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c5b3db1956	Kill off red blobs in most of main/* ... Everything still compiled after making these changes, so I assume these whitespace-only changes didn't break anything (and shouldn't have). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	0e5c761c28	Opaquify ast_channel typedefs, fd arrays, and softhangup flag ... Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fd04da5114	Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. ... When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349729 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	fdda494776	Improve T.38 gateway V.21 preamble detection. ... This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). ........ Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	849992fde9	tweak the v21 detector to detect an additional pattern of hits and misses ... ........ Merged revisions 342605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2e2381341e	Clean up dsp.conf parsing ... Review: https://reviewboard.asterisk.org/r/1434/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	3b1cc6de8d	dsp_process was enhanced to work with alaw and ulaw in addition to slin. ...	14 years ago
Matthew Nicholson	e46aea196c	Merged revisions 328162 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines tune the v21 preamble detector to properly detect the desired sequence of hits and misses ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	3f44b08b7b	do v21 detection instead of CED detection for the fax gateway ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	1339a0a535	Video support for ConfBridge. ... Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1de75f0a4d	Merged revisions 319261 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	c28e2d380c	Merged revisions 317429 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) | 5 lines Only display inband DTMF warning if inband DTMF detection is enabled. (closes issue #18901) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. ... Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	846cfa0ef0	New Feature for chan_dahdi. 4 length pattern matching. ... In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields ... This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	1bc478656e	Set threshold for silence detection defaults to 256 ... (closes issue #15685) Reported by: david_s5 Patches: dsp-silence-threshold-init.diff uploaded by dant (license 670) issue15685.patch.v5 uploaded by pabelanger (license 224) Tested by: danti Review: https://reviewboard.asterisk.org/r/670/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	47ad8c27f5	Fix crash in DTMF detection. ... What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue #17371) Reported by: alecdavis (closes issue #17474) Reported by: kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	07df131a7f	Keep track of digit duration, when we're decoding inband to pass DTMF frames. ... (closes issue #17235) Reported by: frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license 610) 20100518__issue17235.diff.txt uploaded by tilghman (license 14) Tested by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	f55aff74ed	Merged revisions 263949 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	ec0581fef4	restarts busydetector (if enabled) when DTMF is received after call is bridged. ... (closes issue 0016389) Reported by: alecdavis Tested by: alecdavis Patch dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	155931303b	Whitespace. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	6c50fad99f	restarts busydetector (if enabled) when DTMF is received. ... (closes issue #16389) Reported by: alecdavis Tested by: alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthias Nick	71ca1b54cb	Merged revisions 233014 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines Warning message gets displayed only once Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second. (closes issue #15769) Reported by: falves11 Patches: patch_15769_14.txt uploaded by mnick (license 874) Tested by: mnick, falves11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. ... Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	cd10bd931a	Merged revisions 224931 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	bb80c835e0	Add an option, waitfordialtone, for UK analog lines which do not end a call ... until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	455284ebc1	Add a bit of documentation (thanks, I-MOD) on what the silence threshold ... constant actually does and what values are valid for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating ... when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	5b7933fe5e	Janitor patch to change uses of sizeof to ARRAY_LEN ... (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6fd6286a11	arbitrary formatting change to test mantis change ... (closes issue #12824) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause ... - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ae52cd4a76	Merged revisions 114207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines It was possible for a reference to a frame which was part of a freed DSP to still be referenced, leading to memory corruption and eventual crashes. This code change ensures that the dsp is freed when we are finished with the frame. This change is very similar to a change Russell made with translators back a month or so ago. (closes issue #11999) Reported by: destiny6628 Patches: 11999.patch uploaded by putnopvut (license 60) Tested by: destiny6628, victoryure ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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	17 years ago