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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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