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path: root/apps/app_meetme.c
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2007-07-03Fix a build warning, and potential issue if option p is not set at all.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-03Add support for changing the exit key from # to any DTMF.Jason Parker
This does not break existing configs - the arguments to p are optional. Issue 8827, initial patch by junky, mostly rewritten by fw to re-use option p, further modified by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-24Conversions to ast_debug()Russell Bryant
(issue #9984, patches from eliel and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21Merge changes from team/russell/sla_reloadRussell Bryant
* Add support for the reload of sla.conf (closes issue #9481, patch by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21Merged revisions 70726 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) | 2 lines Remove a couple of duplicate unlocks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19Merged revisions 69895 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69895 | file | 2007-06-19 09:55:25 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 lines Perform an extra hangup check just in case. (issue #9589 reported by bcnit) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-15Merged revisions 69518 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) | 5 lines The SLATRUNK_STATUS variable indicated "SUCCESS" for both an answer of the incoming call on the trunk, or if the trunk reached its ring timeout. This patch changes the variable to say "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle, patch by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Convert uses of strdup() to ast_strdup()Russell Bryant
(issue #9983, eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-13Merged revisions 69144 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2 lines Really ignore NULL frames and check whether the channel hungup or not. (issue #9912 reported by junky) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵Tilghman Lesher
guidelines changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-05Merged revisions 67558 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | 5 lines Fix some crashes related to the use of the "meetme" CLI command. The code for this command was not locking the conference list at all. (issue #9351, reported by and patch submitted by Junk-Y, committed patch is different and by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-31Issue #9842 - Doxygen updates by snuffy. Thanks!Olle Johansson
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-26Since this code now uses the API call for creating a detached thread, thereRussell Bryant
is no reason to keep a thread attribute structure on the conference structure. (Pointed out by Tony Mountifield on the asterisk-dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24Add a new API call for creating detached threads. Then, go replace all of theRussell Bryant
places in the code where the same block of code for creating detached threads was replicated. (patch from bbryant) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07Make a minor tweak to admin_exec() - don't lock the conference list until itRussell Bryant
is actually necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07Add a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,Russell Bryant
except it lets you operate on a channel by name instead of conference member number. It is very useful in combination with the 'X' option to ChanSpy. (issue #9671, patch by mnicholson, with some small modifications by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02When a conference is created, the UNIQUEID of the channel that caused it to beRussell Bryant
created will now be stored. Then, every channel that joins the conference will have the MEETMEUNIQUEID channel variable set with this ID. This can be used to relate callers that come and go from long standing conferences. (issue #7295, patch by softins) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-28Enable the functionality of the 'o' option to "optimize talker" by default.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60565 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60565 | russell | 2007-04-06 14:50:52 -0500 (Fri, 06 Apr 2007) | 3 lines When a station picks up a trunk that was on hold, make the hints reflect that nobody has the trunk on hold anymore. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60521 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) | 16 lines Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me) * The original behavior was that if one station put a call on hold, another one picked it up, and then hung up, the code would still consider the call on hold by the first station, so the trunk would not be hung up. However, to better comply with what most people seem to expect it to behave, it will now hang up the trunk. * Fix a problem with "barge=no". This was only intended to prevent people from joining calls that are in progress. However, it also prevented other people from picking up a call that was on hold. This has been fixed. * When there are no active stations on a trunk and it is on hold, the code now indicates the HOLD and UNHOLD conditions to the trunk channel. This allows music on hold to be played to the trunk when it is on hold. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-04Merged revisions 60069 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) | 4 lines Fix a problem where if a trunk was hung up while it was on hold, all of the hints would reflect the line still on hold, even though it should reflect that it is back to not in use. (issue #9459, reported by francesco_r, fixed by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-29Merged revisions 59361 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59361 | file | 2007-03-29 13:38:55 -0400 (Thu, 29 Mar 2007) | 10 lines Merged revisions 59360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 lines Keep a global array of variables indicating whether certain conference rooms are in use. This ensures that two people going into a new dynamic conference when the 'e' option is set don't go into the same conference room. (issue #8835 reported by eliel) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59215 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59215 | russell | 2007-03-26 13:28:29 -0500 (Mon, 26 Mar 2007) | 3 lines Fix some codec negotiation problems when CallerID support is not enabled in SLA. (issue #9308, reported by twilson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14Merged revisions 58894 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines By default, don't attempt to do any CallerID handling at all with SLA because it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-13Merged revisions 58872 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) | 4 lines Ensure that the blinky lights show that the trunk stopped ringing when the trunk hangs up before a station has answered it. (issue #9234, reported by francesco_r) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-10Merged revisions 58669 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) | 2 lines Make the compiler happy and initialize a variable. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08Merged revisions 58512 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | 5 lines Hang up the channel that put the call on hold in the event processing thread to avoid a race condition. Also, if the station originated the call that it is putting on hold, don't hang up the trunk if it was the only station on the call and it is hanging up due to hold and not a normal hangup. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08Merged revisions 58474 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) | 3 lines Refactor hold handling a bit so that it does not require keeping the call up when a call is put on hold. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08Merged revisions 58351-58352 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007) | 2 lines fix two cases where HTTP session file descriptors would not be closed ........ r58352 | kpfleming | 2007-03-08 08:17:42 -0500 (Thu, 08 Mar 2007) | 2 lines fix a compiler warning, and overwriting 'res' value ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-06remove duplicate constLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-05Merged revisions 57872 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 lines Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-05I like it when app_meetme builds under dev mode, don't you?Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-01Merged revisions 57364 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Merged revisions 57203 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines Merge more changes from svn/asterisk/team/russell/sla_updates * Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Merged revisions 57146 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) | 2 lines Minor formatting change ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Merged revisions 57144 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines Merge changes from svn/asterisk/team/russell/sla_updates * Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Merged revisions 57089 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | 8 lines Merge current set of changes from svn/asterisk/team/russell/sla_updates * Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-26Merged revisions 56740 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56740 | russell | 2007-02-25 19:04:40 -0600 (Sun, 25 Feb 2007) | 2 lines Move a comment to be in the correct struct. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Doxygen additions, correctionsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Make Meetme build again under dev mode.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22Merged revisions 56277 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines Merge changes from team/russell/sla_updates. This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55957 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55957 | file | 2007-02-21 15:35:40 -0500 (Wed, 21 Feb 2007) | 10 lines Merged revisions 55956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55951 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55951 | russell | 2007-02-21 14:22:33 -0600 (Wed, 21 Feb 2007) | 3 lines Simplify the last change to app_meetme, and move the call to dispose_conf() up into the block where we know a conf exists. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55949 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55949 | file | 2007-02-21 15:16:34 -0500 (Wed, 21 Feb 2007) | 2 lines Only dispose of the conference if one was created. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55758 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55758 | russell | 2007-02-20 19:03:25 -0600 (Tue, 20 Feb 2007) | 4 lines Improve the reference counting to fix bugs where people report seeing conferences listed that have no members. (issue #9073) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Merged revisions 55006 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55006 | russell | 2007-02-16 16:49:42 -0600 (Fri, 16 Feb 2007) | 17 lines Merged revisions 55005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Merged revisions 54969 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r54969 | russell | 2007-02-16 15:12:18 -0600 (Fri, 16 Feb 2007) | 13 lines Merged revisions 54955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines For conferences that are configured in meetme.conf, check the configuration file every time someone joins the conference instead of only when the conference is first created. This is to ensure that changes to the pin numbers in the config file are always honored. (issue #9073) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-13This introduces a new dialplan function, DEVSTATE, which allows you to do someRussell Bryant
pretty cool things. First, you can get the device state of anything in the dialplan: NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows you to create custom device states so you can control phone lamps directly from the dialplan. Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten => mycustomlamp,hint,Custom:mycustomlamp git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54066 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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