125 Commits (99286417022f0bd43ff35ec5136c312a1fd94e0c)

Author	SHA1	Message	Date
Mark Michelson	9928641702	Fix misuses of asprintf throughout the code. ... This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@371591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	7135c1ccba	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@370643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	e7f158f20c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@369005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	41b01e009f	Allow playback of formats that don't support seeking ... ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	fe36eef8f1	Add Asterisk TestSuite event hooks to support ConfBridge testing ... This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	9a52c30ffa	Video format was treated as audio when removed from the file playback scheduler ... This patch fixes the format type check in ast_closestream and filestream_destructor. Previously a comparison operator was used, but since audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats that have a value greater than the video formats), a bitwise AND operation is used instead. Duplicated code was also moved to filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo Bedrij Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1580/ ........ Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	050f405fbc	Merged revisions 339086 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	be5c67401d	Merged revisions 332817 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	67945ce627	Merged revisions 326209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines Updated filestream destructor to block until move is complete when cache is used When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes. (closes issue ASTERISK-17724) Reported by: Adiren P. Tested by: mjordan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326210 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	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	15 years ago
Sean Bright	50a023add5	Merged revisions 304097 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines Per the man page, setvbuf() must be called before any other operation on an open file. We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2a618dc998	Merged revisions 301446 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines Removal of unused variables so Asterisk will compile. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fad87eea35	Merged revisions 301402 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines Call execl() directly for a better solution for paths with spaces. (closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1d48790cc2	Merged revisions 299989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines Quote arguments, just in case there's a space in a pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by me.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	45432d77b0	Merged revisions 290576 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290576 | tilghman | 2010-10-06 08:49:19 -0500 (Wed, 06 Oct 2010) | 15 lines Merged revisions 290575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) | 8 lines Allow streaming audio from a pipe. (closes issue #18001) Reported by: jamicque Patches: 20100926__issue18001.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	e85f6a3d48	Merged revisions 286270 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, 11 Sep 2010) | 18 lines Merged revisions 286268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines Handle error response when we can't make file compatible Review: https://reviewboard.asterisk.org/r/911/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	5f7c0c349f	Small doxygen fix and doc addition ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3f9c6bb3bc	file.c was truncating audio file formats to the lower 32bits. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a8b0a415fc	Suppress warning in waitstream_core(). ... Suppress the warning about unexpected control subclass frames for AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC in file.c:waitstream_core(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	408ba24fad	Merged revisions 254451 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines Handle new SRCCHANGE control message here too ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	cf6592e58e	Merge tests that verify the same thing. (Oops.) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	962b1a22fd	Try to make ast_format_str_reduce fail... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	98b69d84e1	Merged revisions 238629 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan 2010) | 5 lines Properly calculate the remaining space in the output string when reducing format strings. (closes issue #16560) Reported by: goldwein ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	507e579dc9	Merged revisions 232007 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a warning pointed out by buildbot. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	65c9bfbead	Merged revisions 231740 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	31848bcdd1	Merged revisions 231614 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 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	dd50b9e8b5	Merged revisions 222878 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	07f9778f5b	Merged revisions 219653 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7f745ecd73	Document language prompt submission process. ... This patch adds a document describing the language prompt submission process, licensing terms and other issues related to that process. In addition, it modifies the sound file searching process to support language codes with any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts can be named with gender, customer/company, etc. suffices as well. (closes issue #15771) Reported by: jtodd Patches: language-criteria.txt uploaded by jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	3b91d3b5ab	Revert some silly code that snuck into trunk from my working copy. Sorry! ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. ... Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4021f7d71b	Merged revisions 203785 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	4c0265664e	Merged revisions 200991 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. ... The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	c35e305c82	Fix a memory leak of the write buffer when writing a file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	76a73083a4	Merged revisions 188582 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines Update ast_readvideo_callback to match ast_readaudio_callback. This fixes potential refcount errors that may occur on ast_filestreams. AST-208 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8f28bfc63e	Merged revisions 187300-187301 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b043f8ab1b	Don't act surprised if we get a -1 indication. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	5f7f4a0c84	Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. ... (closes issue #14541) Reported by: grant git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bd244342e2	Fix a problem where file playback would cause fds to remain open forever ... The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b0a8b26ac2	Merged revisions 167566 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines Fix the last couple of places where free() was improperly used directly. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	90431add27	Merged revisions 167554 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines Don't fclose() the file early, the filestream destructor will handle it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9ec93dbcef	Merged revisions 167545 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines Only try to close the file if one was actually opened ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1c7519cd09	Merged revisions 167541 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, #asterisk-dev ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9f7ce9da41	Fix a file playback crash and explicitly initialize values in func_timeout.c ... A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d330d3e210	Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module. ... (closes issue #14079) Reported by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	1e8e12efcf	Janitor, use ARRAY_LEN() when possible. ... (closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago