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
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
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
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
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.
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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
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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
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* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
(patched by me, and tested internally)
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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.).
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r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | 8 lines
Update the SLAStation application to account for the case where the SLA thread
has a call out to the station, but the user has pressed a line button to answer
the call instead of picking up the handset. If they do, the phone sends out a
new INVITE. So, the SLAStation app must check to see if it is picking up a
ringing trunk, and ensure that the other stations stop ringing.
(reported internally, patched by me, tested by mogorman)
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
(closes issue #11078)
Reported by: jthomas
Patches:
meetme-concise.patch uploaded by jthomas (license 293)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines
If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides.
(closes issue #10060)
Reported by: mparker
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menu will adjust this status if a user is muted. The talk request status will
be reflected in the CLI commands as well as the manager interface.
(closes issue #9418)
Reported by: imesper
Patches:
app_meetme_v2.patch uploaded by imesper (license 275)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Reported by: ruffle
Patches:
rb uploaded by ruffle (license 201)
Show whether the conference is locked or not on the CLI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 lines
(closes issue #10122)
Reported by: stevefeinstein
Patches:
meetme-unmute-manager.diff uploaded by qwell (license 4)
Tested by: stevefeinstein
After looking over the code I agree with Qwell. Setting the file descriptor to conference each time just causes a fight back and forth.
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The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) | 7 lines
Fix a problem with the combination of the 'F' option to pass DTMF through a
conference and options that use DTMF to activate various features. The problem
was that the BEGIN frame would be passed through, but the END frame would get
intercepted to activate a feature. Then, the other conference members would hear
DTMF for forever, which they didn't seem to like very much.
(closes issue #10400, reported by stevefeinstein, fixed by me)
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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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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.
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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)
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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)
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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)
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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)
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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.
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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)
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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
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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)
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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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.
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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
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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.
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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
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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)
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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
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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
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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the location of the header files.
On passing, add a cast to insure -Werror clean compilation
on FreeBSD 6.x, where time_t does not match %ld
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r42783 | tilghman | 2006-09-11 16:47:23 -0500 (Mon, 11 Sep 2006) | 4 lines
When paging, only wait 5 seconds for the marked user to enter the conference.
After that, assume the paging already completed by the time the channel entered
the conference and drop back out. (Issue 7275)
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- use ast_calloc instead of malloc + memset
- return immediately on ast_calloc failure instead of indenting the whole func
- remove a duplicate ast_strdupa
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-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
support the new location for zaptel.h and tonezone.h
use the dependency information output by menuselect to build Makefile rules for each module for header files and libraries
combine the common rules into a top-level Makefile.rules file
remove all (now) unnecessary stuff from subdir Makefiles
change translator API so that the newpvt() callback returns an int instead of a pointer (it no longer allocates memory)
alphabetize --with-<foo> options in configure script
enhance Net-SNMP support in configure script to provide a --with-netsnmp option
fix support for --with-pq so that if pg-config is not found when --with-pq is specified, an error will be generated
add 'optional package' usage to modules now that menuselect can output it
allow res_snmp to build by default, since the new loader changes coming soon will solve the function naming problem (and users can disable it via menuselect anyway)
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As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely. Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.
Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.
I am just sorry that this change missed SVN version number 20000!
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r18087 | kpfleming | 2006-04-06 17:04:58 -0500 (Thu, 06 Apr 2006) | 3 lines
don't try to support 'i' or 'r' options if chan_zap is not loaded, and warn the user when they attempt to use them (issue #6675)
update application help text to more clearly define when Zaptel and chan_zap are required
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in pbx_exec is always 1 so it can be removed.
This change also takes away ast_exec_extension(), and lets all
switch functions (exists, canmatch, exec, matchmore) all use the same
prototype, which makes the code a bit cleaner.
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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
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allows for meetme show list to tell how long user
has been in confrence and reports it to manager
when user leaves confrence.
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This should prevent us from unintentionally changing variable
values when they're returned from pbx_builtin_getvar_helper.
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In general, LOCAL_USER_ADD/REMOVE should be the first/last thing called in an
application. An exception is if there is some *fast* setup code that might
halt the execution of the application, such as checking to see if an argument
exists.
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Mark Michelson
Sean Bright
Michiel van Baak
Tilghman Lesher
Kevin P. Fleming
Brett Bryant
Jeff Peeler
Joshua Colp
Russell Bryant
Terry Wilson
Steve Murphy
Jason Parker
Olle Johansson
Luigi Rizzo
Dwayne M. Hubbard
Matt O'Gorman
Mark Spencer
BJ Weschke
Josh Roberson