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path: root/channels/chan_sip.c
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2007-11-16remove redundant #include "asterisk/compat.h",Luigi Rizzo
but make sure that asterisk/compiler.h is included everywhere git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
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
2007-11-15And file said... let trunk build again! Accomplished by some more ↵Joshua Colp
constification, and marking a function in chan_sip as purposely unused until it is fixed up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Always relying on the responses when crossing NAT's are not a goodOlle Johansson
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: https://origsvn.digium.com/svn/asterisk/trunk@89285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Merged revisions 89281 via svnmerge from Olle Johansson
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: https://origsvn.digium.com/svn/asterisk/trunk@89283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Merged revisions 89280 via svnmerge from Olle Johansson
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: https://origsvn.digium.com/svn/asterisk/trunk@89282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Exit early instead of deciding to exit after processing the message.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Add support for application/dtmf SIP INFO dtmf handling. Yep, anotherOlle Johansson
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: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14make the 'name' and 'value' fields in ast_variable const char *Luigi Rizzo
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: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13- Convert initialization of a struct to C99 style instead of GNU styleRussell Bryant
- Fix a minor spelling error in a comment git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13Merged revisions 89246 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@89247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89184 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@89185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89119 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@89120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89101 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@89102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89099 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@89100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89097 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@89098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Merged revisions 89095 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@89096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Merged revisions 89090 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@89091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Commit some cleanups to the format type code.Tilghman Lesher
- 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: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06"show application <foo>" changes for clarity.Mark Michelson
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Merged revisions 89032 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@89034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88768 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@88769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88765 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines Fix up some indentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88671 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88585 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config structure on configuration failure. Issue 11163, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-04Simplify the implementation and the API for stringfields;Luigi Rizzo
details and examples are in include/asterisk/stringfields.h. Not applicable to older branches except for 1.4 which will receive a fix for the routines that free memory pools. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merged revisions 88366 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merged revisions 88328 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01Change some uses of free() to ast_free(). (No functional differences.)Russell Bryant
(closes issue #11138) Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-29Merged revisions 87342 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. (closes issue #9566) Reported by: atca_pres Patches: bug9566.patch uploaded by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵Jason Parker
didn't make much sense git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22Merged revisions 86756 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 lines After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. (closes issue #10113) Reported by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19Convert NEW_CLI to AST_CLI.Jason Parker
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19Merged revisions 86471 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote. (closes issue #11027) Reported by: ramonpeek Patches: 11027-1.diff uploaded by ramonpeek (license 266) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19Merged revisions 86469 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 lines Set port number in received as information for registrations as well. (closes issue #11028) Reported by: brad-x ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-17Merged revisions 86149 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines If Asterisk is in the middle of shutting down, respond to OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: eserra Patches: sip-options-503.patch uploaded by eserra (license 45) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-17Merged revisions 86117 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4 lines Whoops, forgot to remove the original sip_scheddestroy. (closes issue #11010) Reported by: vadim ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-17Merged revisions 86063 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4 lines Don't schedule dialog destruction if a MESSAGE is received using an existing dialog. (closes issue #11010) Reported by: vadim ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-16This fixes SIP subscriptions in trunk.Russell Bryant
Don't improperly memset() over an ast_str. This was leftover from before it got changed to use ast_str. (closes issue #11003, reported by pj) (closes issue #10770, reported by yehavi) (patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-15Switch dundi to new tos config format.Jason Parker
Remove old unused defines for old style. Closes issue 10860, patch by IgorG. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-15Merged revisions 85604 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines Make the default for the srvlookup option to be yes. It doesn't really make sense for it to default to off. The default configuration file has it on, and proper RFC behavior, as indicated by a comment in the code, is for it to be on. So, let's have it on by default to make lives easier. (closes issue #10954, suggested by jtodd) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-10Merged revisions 85280 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4 lines If devicestate is passed a port number strip it out. (closes issue #10930) Reported by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-10Merged revisions 85277 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6 lines Add support for handling a 182 Queued response. (closes issue #10924) Reported by: ramonpeek Patches: queued-182.diff uploaded by ramonpeek (license 266) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09Merged revisions 85093 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4 lines Don't perform a reinvite if a transfer is in progress. (issue #10915) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-02Merged revisions 84370 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | 6 lines Use snprintf instead of sprintf in one place. There is no vulnerability here due to various buffer sizes around the code, but I still didn't like seeing a non length-limited copy of data coming off of the wire into a stack buffer, as this would be a problem in the future if buffer sizes elsewhere got changed or size limitations removed ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01Check to make sure a structure pointer is non-NULL before touching it... ↵Joshua Colp
crashing is bad, mmmk? (closes issue #10831) Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01Corydon posted this janitor project to the bug tracker and mvanbaak providedRussell Bryant
a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string (closes issue #10843) Reported by: Corydon76 Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01Add MP4 to part of the SDP code.Joshua Colp
(closes issue #10820) Reported by: ruikubo Patches: chan_sip.patch uploaded by ruikubo (license 250) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-26Merged revisions 83943 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) | 2 lines I changed my mind ... I think this should be a LOG_NOTICE. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-26Merged revisions 83941 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | 5 lines Add a log message that was requested by the masses in the developer tutorial session at Astricon. chan_sip did not output any message when a call was rejected because the extension was not found. This adds a verbose message (at verbose level 3) to note when this happens. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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