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2009-05-22Implement a new element in AstXML for AMI actions documentation.Eliel C. Sardanons
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: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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
2009-05-20Merged revisions 195635 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@195636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Fix a bug where the 'T' option to Meetme did not work.Joshua Colp
(closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-12add 'const' qualifiers in various places where they should have beenKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03app_meetme not setting filename and fileformat correctly for realtimeDavid Vossel
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: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Merged revisions 179532 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18Re-add 'o' option to MeetMe, reverting rev 62297.Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@177101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 176249,176252 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@176253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22Merged revisions 170147 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Add a missing unlock and properly handle the 'maxusers' setting on MeetMeSean Bright
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: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Fix the buildMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Merged revisions 165255 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@165318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-09Merged revisions 162286 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@162291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-18Merged revisions 157365 via svnmerge from Jeff Peeler
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: https://origsvn.digium.com/svn/asterisk/trunk@157366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-13Introduce XML documentation for:Eliel C. Sardanons
- 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: https://origsvn.digium.com/svn/asterisk/trunk@156575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Merged revisions 156294 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@156295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Merged revisions 156289 via svnmerge from Jeff Peeler
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: https://origsvn.digium.com/svn/asterisk/trunk@156290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Merged revisions 156178 via svnmerge from Jeff Peeler
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: https://origsvn.digium.com/svn/asterisk/trunk@156228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Fix option handling code.Tilghman Lesher
(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: https://origsvn.digium.com/svn/asterisk/trunk@150384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Initialize character arrays as they are not guaranteed to be set.Jeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-08Some small tweaks regarding realtime conference announcements.Mark Michelson
(closes issue #13522) Reported by: DEA Patches: meetme-rt-fixes.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-08Keep up with shadow warnings. One day I'll actually enable this in the ↵Sean Bright
Makefile. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02fix the 'meetme list', 'meetme list concise', 'meetme list $confno' and ↵Michiel van Baak
'meetme list $confno concise' CLI commands (closes issue #13586) Reported by: john8675309 Help and feedback from eliel, thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02make this compile under devmode againMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01Add schedule extensions to app_meetme. In addition, the reporter found aTilghman Lesher
problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-28Merge the cli_cleanup branch.Michiel van Baak
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
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingTilghman Lesher
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
2008-08-10More RSW merges. Everything from apps/ except for the big offendersSean Bright
app_voicemail and app_queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-06Janitor ast_str projectTilghman Lesher
(closes issue #13058) Reported by: pputman Patches: app_meetme_aststr2.patch uploaded by pputman (license 81) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-29build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for ↵Kevin P. Fleming
features that will always be present in DAHDI git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28merging the zap_and_dahdi_trunk branch up to trunkMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08Janitor project to convert sizeof to ARRAY_LEN macro.Brett Bryant
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-25Merged revisions 125132 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵Jeff Peeler
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Merged revisions 122137 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines Flipflop the sections for two options, since the section for 'X' (exit context) may otherwise absorb keypresses meant for 's' (admin/user menu). (closes issue #12836) Reported by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-09Expand RQ_INTEGER type out to multiple types, one for each precisionTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05Merge the adaptive realtime branch, which will make adding new required fieldsTilghman Lesher
to realtime less painful in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causeMichiel van Baak
- 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
2008-04-30Lock around variables retrieved, and copy the values, if they stay persistent,Tilghman Lesher
since another thread could remove them. (closes issue #12541) Reported by: snuffy Patches: bug_12156_apps.diff uploaded by snuffy (license 35) Several additional changes by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Fix DST calculation, and fix bug in calculation of whether conf has started ↵Tilghman Lesher
yet or not (Closes issue #12292) Reported by: DEA Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18Add MEETME_INFO dialplan function that allows querying various properties of ↵Joshua Colp
a Meetme conference. (closes issue #11691) Reported by: junky Patches: meetme_info.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-13Use ast_mkdir instead of mkdirTilghman Lesher
(Closes issue #12430) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-10Merged revisions 114029 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines Create the directory where name recordings will go if it does not exist. (closes issue #12311) Reported by: rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20Remove astobj.h from some places where it wasn't neededRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19Merged revisions 110163 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines Fix a bug where when calls on the trunk side hang up while on hold, the state is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18Go through and fix a bunch of places where character strings were being ↵Terry Wilson
interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11Merged revisions 107637 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened. (closes issue #12136) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07Merged revisions 106895 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines Only start the SLA thread if SLA has actually been configured. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-05 23:04:49 +0000