58 Commits (0ae9097e3e7326bbc0c4b1ec0c6dd8e2127d7a2e)

Author	SHA1	Message	Date
Russell Bryant	405d6cdf31	Add support for devices with less than 3 lines on the LCD. ... (closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. ... This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	862ebf4d00	fixes adaptive jitterbuffer configuration ... 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	16 years ago
Tilghman Lesher	f59fe83c56	More 32->64 bit codec conversions. ... In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 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	0ff53eddef	Always specify which RTP engine is desired for a new RTP instance. ... This fixes a crash reported in #asterisk-dev where chan_mgcp unexpectedly allocated an RTP instance from res_rtp_multicast, since by not specifying an engine, you get the first one in the list of engines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... 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	17 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	17 years ago
Tilghman Lesher	e4149506a1	Send DTMF frame before playing back audio. ... (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. ... There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9cbd90a864	Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. ... (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	8e4b5df187	Fix some uninitialized memory notices that appeared under valgrind. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. ... The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. ... This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ee77b475f2	Improve performance of the ast_event cache functionality. ... This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	115c6abef4	Merged revisions 175921 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	90751b16ca	Merge the cli_cleanup branch. ... 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	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
Sean Bright	db1ed285c4	More RSW merges. This should do it for the channels/ dir. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	28534ea921	Swap "static" and "const", so that "static" appears at the beginning of each ... declaration (suppresses a warning). (closes issue #13070) Reported by: gknispel_proformatique Patches: asterisk_trunk_const_static.patch uploaded by gknispel (license 261) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	885d17506b	Keep ast_app_inboxcount API compatible with 1.6.0. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 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	18 years ago
Mark Michelson	7daebcd610	Adding support for "urgent" voicemail messages. Messages which are ... marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	08e674bce0	Pass the hangup cause all the way to the calling app/channel. ... (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	18 years ago
Jeff Peeler	41fd7a6a21	(closes issue #6113) ... Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	b05e17fdd7	Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. ... (closes issue #12460) Reported by: PerryB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b9bb0749d1	Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone). ... (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	4a21c5dd22	Fix spelling of existent in a few places. ... (closes issue #12409) Reported by: candlerb git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 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
Mark Michelson	fe9821cc10	Get rid of any remaining ast_verbose calls in the code in favor of ... ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fd88390af7	remove unnecessary (char *) casts for ast_config_AST_* variables. ... There are some left in the .flex files, left to the maintainer... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	acd40b76a3	You... will... build! I say so and therefore you will. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	98307c16ed	minor cleanups ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	535705ca86	fix building under cygwin. ... At this point WINARCH should go away. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	0b34b8e56c	remove some unnecessary includes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	01f8a6959b	make the configure script detect that it is running on a Windows platform, 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	18 years ago
Joshua Colp	97b3e7ec72	If no timezone is available use the default message. ... (closes issue #11576) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6267956bda	Make chan_unistim actually be able to unload. When creating a thread that you want to pthread_join you have to explicitly create it as joinable, and also if using pthread_cancel you have to have a pthread_testcancel to see if it has been called. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	17afebc1a6	HUGE improvements to QoS/CoS handling by IgorG ... - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7029e8fdd1	Solaris compat fixes ... Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11315) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	7e8835e0d7	remove another set of redundant #include "asterisk/options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	6d8d66e9e7	Fix some code that was supposed to ensure that a buffer was terminated, but was ... writing to the wrong byte. Also, remove some non-thread safe test code. (closes issue #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by IgorG (license 20) - additional changes by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	a23c055c3d	move asterisk/paths.h outside asterisk.h and into those files ... who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	0595b5e2aa	include "logger.h" and errno.h from asterisk.h - usage shows that they ... were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	d82a631f9c	more removal of duplicate #include lines ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	cfbc4f5fb8	Doxygen fixes. ... Also fix a common typo I kept seeing (arguement) in various files. Closes issue #11222, patch by snuffy (with arguement > argument by me). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	098728eb64	Fix building on FreeBSD by including/not including some headers. ... (closes issue #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago