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2007-11-16remove redundant #include "asterisk/compat.h",rizzo1-1/+0
but make sure that asterisk/compiler.h is included everywhere git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89336 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo22-94/+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
2007-11-16fixed #10631, about one way audio. thanks IgorG again.crichter1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89321 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16move the inner part of config file parsing to a separate function,rizzo1-22/+27
so it can be reused in the implementation of cli commands when they have a similar syntax. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89320 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16fixed compilation of chan_misdn, #11269, thanks IgorG.crichter1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89319 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Merged revisions 89301 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 Nov 2007) | 2 lines Fix an uninitialized memory read found by valgrind ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89303 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Merged revisions 89298 via svnmerge from tilghman1-0/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) | 5 lines Yet another memory corruption issue. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89299 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15And file said... let trunk build again! Accomplished by some more ↵file3-8/+9
constification, and marking a function in chan_sip as purposely unused until it is fixed up. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89290 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Always relying on the responses when crossing NAT's are not a goodoej1-1/+7
solution, it breaks communication. Rizzo - you need to implement a configuration option for this code. It's good, but maybe should be off by default. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89285 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Merged revisions 89281 via svnmerge from oej1-5/+17
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines Don't send re-invites during pending INVITE transactions. Patch by one47 - thanks! Closes issue #9305 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89283 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Merged revisions 89280 via svnmerge from oej1-18/+25
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines Improve support for multipart messages. Code by gasparz, changes by me (mostly formatting). Thanks, gasparz! Closes issue #10947 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89282 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Exit early instead of deciding to exit after processing the message.oej1-9/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89279 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Add support for application/dtmf SIP INFO dtmf handling. Yep, anotheroej1-11/+82
way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89278 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14One more typo in config.c; and missed conversions due to the constifying of ↵tilghman2-5/+5
ast_variable_new parameters git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89270 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14make the 'name' and 'value' fields in ast_variable const char *rizzo5-16/+21
This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89268 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13- Convert initialization of a struct to C99 style instead of GNU stylerussell1-7/+7
- Fix a minor spelling error in a comment git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89251 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Merged revisions 89246 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines If we set a value for qualify, we should actually pay attention to it, instead of overriding the value ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89247 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Doxygen fixes.qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@89202 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89184 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines Fix two cases of memory corruption caused by background threads. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89185 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89173 via svnmerge from crichter1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89179 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89172 via svnmerge from crichter4-24/+41
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89178 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Fix building on FreeBSD by including/not including some headers.file1-0/+1
(closes issue #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys (license 281) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89177 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89171 via svnmerge from crichter2-31/+44
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89176 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89170 via svnmerge from crichter1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 Nov 2007) | 1 line fixed the support for CW and therefore for the reject_cause option. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89175 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Merged revisions 89169 via svnmerge from crichter5-2/+27
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89174 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Merged revisions 89119 via svnmerge from mmichelson1-27/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89120 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08convert this code to a more efficient idiomkpfleming1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89118 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Fix missed conversion to linkedlists macro changetilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89113 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08initialize a variable to silence compiler.rizzo1-1/+1
The type of warnings emitted depends on the optimization level, at the lower levels the compiler doesn't always understand what the programmer has in mind. In this case I could not understand it either. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89108 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08improve linked-list macros in two ways:kpfleming3-31/+22
- the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Merged revisions 89101 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue #10756) Reported by: goestelecom ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89102 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Merged revisions 89099 via svnmerge from file1-6/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89100 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08Merged revisions 89097 via svnmerge from file1-2/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89098 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Merged revisions 89095 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue #11185) Reported by: spditner ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89096 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Merged revisions 89090 via svnmerge from mmichelson1-2/+24
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89091 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Minor change so chan_h323 builds again.file1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89086 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Provide the ability to directly manipulate the TON/NPI bits in the dialstring.tilghman1-1/+102
Reported by: thetatag Patch by: thetatag/stevens/tilghman Closes issue #5331 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89078 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Commit some cleanups to the format type code.tilghman7-20/+15
- 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
2007-11-06Add some more locking as well as API update for libss7 for new transport typesmattf1-2/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89067 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06"show application <foo>" changes for clarity.mmichelson2-5/+5
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Allow gtalk and jingle to use TLS connections again.qwell2-14/+14
Closes issue #9972 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89041 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Merged revisions 89032 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue #11006) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89034 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Merged revisions 88994 via svnmerge from file1-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88995 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Update chan_agent documentation. Change a | to , as that is now the required ↵file1-1/+1
way. (closes issue #11167) Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88974 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Set up detection of IP_PKTINFO in autoconf for chan_unistimtilghman1-4/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88973 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06convert uses of LOG_DEBUG to use ast_debug()russell1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88937 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Add jitterbuffer support to chan_unistim.russell1-1/+22
(closes issue #11168) Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88935 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Merged revisions 88805 via svnmerge from russell2-36/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines After seeing crashes related to channel variables, I went looking around at the ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88934 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06explain that the host environment must be used to build gentone;rizzo1-5/+5
Remove unset variables, they would be misleading. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Merged revisions 88768 via svnmerge from russell1-2/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines When traversing the list of channel variables here in transmit_invite(), the asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88769 f38db490-d61c-443f-a65b-d21fe96a405b
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