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2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler1-3/+158
in-conf This patch adds the following in-conference admin DTMF features: *81 - Roll call (or simply user count if INTROUSER isn't enabled) *82 - Eject all non-admins *83 - Mute/unmute all non-admins *84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Export MEETMEBOOKID and fix pin-less conferences with realtime conferencesrussell1-0/+13
(closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256019 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Resolve a crash in SLATrunk when the specified trunk doesn't exist.seanbright1-1/+0
Reported by philipp64 in #asterisk-dev. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252623 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-15Convert a few places to use ast_calloc_with_stringfields where applicable.seanbright1-8/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240368 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Fix misreverting from 177158.jpeeler1-1/+1
(closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238181 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Merged revisions 236509 via svnmerge from seanbright1-30/+34
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-11Merged revisions 234379 via svnmerge from jpeeler1-20/+37
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines Fix talking detection status after conference user is muted. This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234380 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Add audio announcement option to app_pagejpeeler1-124/+157
As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234173 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Display a list of channel variables in each channel-oriented event.tilghman1-8/+8
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Yet another error message in the dialplan (thanks, rmudgett/russellb)tilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228196 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05MEETME_INFO should not return a literal error message to the dialplan.tilghman1-1/+2
(closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228191 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-9/+9
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225105 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225360 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Apparently, I don't need to specify the ".so" suffix to get a matchtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225102 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Turn on DENOISE filter for all conference participants.tilghman1-1/+7
(Fixes SWP-238) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225048 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Fix compilation of app_meetme.seanbright1-1/+1
Reported by ebroad in #asterisk-bugs git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217286 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Merged revisions 217156 via svnmerge from tilghman1-9/+25
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Small doxygen changesoej1-12/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212844 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-15/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-14Document all meetme realtime fields, and in the process, make some field ↵tilghman1-8/+8
lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206567 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29A few const changes in app_meetme.c that I noticed while browsing the source.seanbright1-5/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.russell1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Merged revisions 200991 via svnmerge from kpfleming1-3/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-06Move function MEETME_INFO documentation to XML.eliel1-9/+34
Move function MEETME_INFO static documentation to the new AstXML form. (issue #15245) Reported by: eliel Patches: app_meetme_static_conversion.txt uploaded by lmadsen (license 10) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199409 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Implement a new element in AstXML for AMI actions documentation.eliel1-14/+43
A new xml element was created to manage the AMI actions documentation, using AstXML. To register a manager action using XML documentation it is now possible using ast_manager_register_xml(). The CLI command 'manager show command' can be used to show the parsed documentation. Example manager xml documentation: <manager name="ami action name" language="en_US"> <synopsis> AMI action synopsis. </synopsis> <syntax> <xi:include xpointer="xpointer(...)" /> <-- for ActionID <parameter name="header1" required="true"> <para>Description</para> </parameter> ... </syntax> <description> <para>AMI action description</para> </description> <see-also> ... </see-also> </manager> git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-11/+12
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: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-20Merged revisions 195635 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195636 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-14Fix a bug where the 'T' option to Meetme did not work.file1-2/+2
(closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194434 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-12add 'const' qualifiers in various places where they should have beenkpfleming1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193832 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03app_meetme not setting filename and fileformat correctly for realtimedvossel1-3/+39
When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179972 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179532 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179533 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Re-add 'o' option to MeetMe, reverting rev 62297.russell1-6/+10
Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177101 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176249,176252 via svnmerge from mmichelson1-14/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176253 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-22Merged revisions 170147 via svnmerge from file1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@170148 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Add a missing unlock and properly handle the 'maxusers' setting on MeetMeseanbright1-1/+3
conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168705 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Fix the buildmmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165326 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165255 via svnmerge from mmichelson1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165318 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)tilghman1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163991 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162286 via svnmerge from russell1-7/+27
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162291 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-18Merged revisions 157365 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157366 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-13Introduce XML documentation for:eliel1-149/+365
- MeetMe() - MeetMeCount() - MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk() - Add an attribute to optionlist 'hasparams' with the same functionality as the one we have in <parameter> and <argument> (the DTD was updated) - Fix a leak when getting an attribute while parsing an <optionlist>. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156575 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12Merged revisions 156294 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156295 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12Merged revisions 156289 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156290 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12Merged revisions 156178 via svnmerge from jpeeler1-24/+148
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156228 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Fix option handling code.tilghman1-13/+26
(closes issue #11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt uploaded by DEA (license 3) with additional fixes by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150384 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Initialize character arrays as they are not guaranteed to be set.jpeeler1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150309 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-08Some small tweaks regarding realtime conference announcements.mmichelson1-3/+3
(closes issue #13522) Reported by: DEA Patches: meetme-rt-fixes.txt uploaded by DEA (license 3) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147714 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-08Keep up with shadow warnings. One day I'll actually enable this in the ↵seanbright1-6/+6
Makefile. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147457 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-02fix the 'meetme list', 'meetme list concise', 'meetme list $confno' and ↵mvanbaak1-70/+66
'meetme list $confno concise' CLI commands (closes issue #13586) Reported by: john8675309 Help and feedback from eliel, thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145915 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-02make this compile under devmode againmvanbaak1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145842 f38db490-d61c-443f-a65b-d21fe96a405b
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