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path: root/channels/chan_oss.c
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2012-08-21Fix misuses of asprintf throughout the code.Mark Michelson
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 ........ Merged revisions 371591 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-24chan_oss: fix "sample rate" error messageTzafrir Cohen
Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged revisions 370432 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-14Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)Kinsey Moore
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-08Eliminate a bunch of shadow warnings.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29Opaquify ast_channel structs and listsTerry Wilson
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24Opaquification for ast_format structs in struct ast_channelTerry Wilson
Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20ast_channel opaquification of pointers and integral typesTerry Wilson
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-13Opaquify char * and char[] in ast_channelTerry Wilson
Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-01Constify some more channel driver technology callback parameters.Richard Mudgett
Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Opaquify channel stringfieldsTerry Wilson
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
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
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-05Merged revisions 317478 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
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
2010-08-15Support for GNU/kFreeBSDTzafrir Cohen
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING This patch gets Asterisk close to building on Debian kFreeBSD i386, mainly by adding an extra test for __GLIBC__ in one or two (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support (and support for chan_vpb) was not tested. Review: https://reviewboard.asterisk.org/r/858/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Expand the caller ANI field to an ast_party_idRichard Mudgett
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14ast_callerid restructuringRichard Mudgett
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
2010-06-09Merged revisions 269495 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.Richard Mudgett
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02fixes adaptive jitterbuffer configurationDavid Vossel
When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-12Fix mute toggling on OSS channels.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10AST-2009-005Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28Define side-effect-safe MIN and MAX macros and remove duplicate definitions ↵Kevin P. Fleming
from various files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.Russell Bryant
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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
2009-01-25Merged revisions 171187 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Numerous documentation updates.Joshua Colp
(closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Janitor, use ARRAY_LEN() when possible.Eliel C. Sardanons
(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
2008-11-04improve configure script to remember the previous value of each dependency ↵Kevin P. Fleming
in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 ↵Kevin P. Fleming
branch, and add the ones needed for all the new code here too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09(closes issue #13557)Steve Murphy
Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-28Merge the cli_cleanup branch.Michiel van Baak
This work is done by lmadsen, junky and mvanbaak during AstriDevCon. This is the second audit the CLI got, and this time lmadsen made sure he had _ALL_ modules loaded that have CLI commands in them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-25(closes issue #13557)Steve Murphy
Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingTilghman Lesher
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
2008-09-09Fix a memory leak in chan_ossMark Michelson
(closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-30implement the 'freeze' function for incoming frames;Luigi Rizzo
fix a bug which caused a crash when a videodevice was specified after startgui=1 in the config file. This also involves a slightly different method to determine if the gui is active or not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-29implement a 'toggle' option for 'console mute' and 'console unmute'Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causeMichiel van Baak
- 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
2008-04-24Pass the hangup cause all the way to the calling app/channel.Michiel van Baak
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-14Merged revisions 108796 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Merged revisions 106235 via svnmerge from Joshua Colp
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
2008-01-09make get_video_desc() return the active console ifLuigi Rizzo
passed a null argument (channel). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-03eliminiate sound_thread() and other stuff from chan_oss since Asterisk ↵Kevin P. Fleming
indications can handle it remove gentone and all the headers containing tones that are no longer needed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-27remove useless castsLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-22Build console_video support by linking in, as opposed to including,Luigi Rizzo
console_video.c This will ease the task of splitting console_video.c into its components (V4L and X11 grabbers, various video codecs and packetizers, SDL), as well as ease future extensions (e.g. additional video sources, codecs and rendering engines). For the time being nothing changes for users: video support is off by default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included (if SDL and FFMPEG are available). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-20add some macros to simplify parsing the config file,Luigi Rizzo
see description in config.h . They are a variant of the set of macros i used in chan_oss.c, structured in a way to be more robust to the presence of spurious ';' - basically, they define wrappers for 'do {' and '} while (0)', plus some helper functions to deal with simple cases such as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix (CV_ as 'Config Variable') tries to be easy to remember and has been chosen to not conflict with other existing macros in the tree. For the time being, I have only updated the three source files in the tree that used the old M_* macros. Hopefully, more files will be converted. NOTE: I understand that inventing my own dialect of C is generally wrong; however, the lack of adequate support in the language encourages lazy programming practices (such as ignoring errors, bounds, etc.) and this increases the chance of vulnerability in the code, especially because we are parsing user input here. Hopefully, these macros and the use of ast_parse_arg (in config.h) should encourage the programmer to write more robust code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-17surprising as it may be, chan_oss compiles correctly under cygwin as well,Luigi Rizzo
provided you look for soundcard.h in the right place... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-17make the configure script detect that it is running on a Windows platform, ↵Kevin P. Fleming
and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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