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2009-04-24Convert the ast_channel data structure over to the astobj2 framework.russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Fix some uninitialized memory notices that appeared under valgrind.file1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187772 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merge in the RTP engine API.file1-49/+46
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: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185362 via svnmerge from dbrooks1-7/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Issue a warning message if our candidate's IP is the loopback address.phsultan1-0/+3
(closes issue #13985) Reported by: jcovert Tested by: phsultan git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175089 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175029 via svnmerge from phsultan1-5/+47
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175058 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-21Fix two memory leaks in chan_gtalk, thanks Eliel!phsultan1-0/+2
(closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139281 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Janitor patch to change uses of sizeof to ARRAY_LENbbryant1-2/+2
(closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-02Do not link the guest account with any configured XMPP client (inphsultan1-10/+26
jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119741 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-23- remove whitespaces between tags in received XML packets before givingphsultan1-1/+4
them to the parser ; - report Gtalk error messages from a buddy to the console. This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation work with Empathy. Note that this is only true for audio streams, not video. Thank you to PH for his great help! (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff uploaded by phsultan (license 73) Tested by: phsultan, PH git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118020 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak1-2/+2
- 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: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Merged revisions 114708 via svnmerge from tilghman1-2/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114709 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-24Pass the hangup cause all the way to the calling app/channel.mvanbaak1-2/+2
(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: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-0/+12
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: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-04Merged revisions 112820 via svnmerge from phsultan1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112821 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-04Merged revisions 112766 via svnmerge from phsultan1-45/+124
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112785 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107714 via svnmerge from qwell1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber). (closes issue #12014) Reported by: junky ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107718 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29Remove unnecessary if statements before calling iks_delete (redundant check isphsultan1-46/+34
done inside iks_delete), thus making the code conform with coding guidelines. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105263 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97489 via svnmerge from phsultan1-15/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97490 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Manager events from the "moremanager" branchoej1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-17more removal of duplicate #include linesrizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89349 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-5/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Commit some cleanups to the format type code.tilghman1-1/+1
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Allow gtalk and jingle to use TLS connections again.qwell1-7/+7
Closes issue #9972 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89041 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Remove traces of gnutls, since we no longer use/need it.qwell1-10/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Merged revisions 87906 via svnmerge from qwell1-3/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11130) (closes issue #11132) ........ r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try to allocate memory that we're just going to re-allocate later anyways. Issues 11130 and 11132, patch by eliel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87907 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-2/+2
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-2/+2
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Fix CLI help outputphsultan1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85787 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)russell1-28/+38
(closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-13Merged revisions 82309 via svnmerge from phsultan1-13/+29
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 Sep 2007) | 4 lines Closes issue #9401, reported and patched by irrot, with slight modifications by me. Handle DTMF sent by Asterisk properly. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82310 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-06Merged revisions 81743 via svnmerge from phsultan1-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) | 1 line Various string length fixes. Removed an unused variable in aji_client structure (context) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81745 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-31Merged revisions 81410 via svnmerge from phsultan1-2/+43
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command available. Closes issue 10548, reported by keepitcool. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81411 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-24Merged revisions 80661 via svnmerge from phsultan1-1/+27
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are answered too early, which results in CDRs wrongly stating that a call was ANSWERED when the calling party cancelled a call before before being established. We must not answer the call upon reception of a 'transport-accept' iq packet, but this packet still needs to be acknowledged, otherwise the remote peer would close the call (like in #8970). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80662 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Merged revisions 79174 via svnmerge from file1-2/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79175 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Add support for using epoll instead of poll. This should increase ↵file1-4/+4
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Merged revisions 72331 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72332 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Merged revisions 72125 via svnmerge from qwell1-7/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines Don't modify a variable that we don't want modified. Make a copy of it instead. Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). Note: chan_jingle in trunk does not appear to have the same bug. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72134 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Merged revisions 70084 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines Only attempt to queue a hangup on the owner channel if it actually exists. (issue #9795, patch from zandbelt) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70088 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.russell1-19/+11
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵tilghman1-5/+5
guidelines changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-03ast_calloc janitor (Inspired by issue 9860)tilghman1-5/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66981 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-25Merged revisions 66157 via svnmerge from kpfleming1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines handle the GNUTLS library properly in the configure script and build system don't build in OSP support unless we have found and are allowed to use SSL support ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66158 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Merged revisions 65965-65967 via svnmerge from kpfleming1-1/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines don't use uninitialized variables ........ r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines don't reference GnuTLS headers and functions unless the configure script found it ........ r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use #ifdef instead of #if ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65979 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Merged revisions 65901 via svnmerge from oej1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65904 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Merged revisions 65892 via svnmerge from oej1-2/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65898 f38db490-d61c-443f-a65b-d21fe96a405b
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