option is incorrectly passed to the transferee when built-in
attended transfers are used. There is still a problem with 'T',
but better to fix some problems than no problems while we work
on it.
(closes issue #7904)
Reported by: k-egg
Patches:
transfer-fix-trunk-r97657.diff uploaded by sergee (license 138)
Tested by: sergee, otherwiseguy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines
If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else.
(closes issue #11743)
Reported by: davevg
Patches:
res_agi.diff uploaded by davevg (license 209)
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r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line
Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro.
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the XMPP over TLS/SSL connection resets for some reason, it is
wrongly believed as being secured, which makes the re-connection
process endlessly fail. This was reported by mvanbaak in issue #11644.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that paths and filename are writable by asterisk.c without
causing segfaults.
This involves defining the variables as const char *,
and having them point to as static, writable buffer
defined in asterisk.c
On passing, fix some errors in using these variables
in some files in utils/ , and in res/snmp/agent.c
which was redefining a variable without using paths.h
(not applicable to 1.4)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec 2007) | 6 lines
Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor
needs to use this if the user does not have soxmix.
(closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me)
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is below. I should note that the PACKAGE_* macros that asterisk
defines in autoconfig.h are not used anywhere in the tree so
they should just be removed.
/*
* There is some collision collision between netsmp and asterisk names,
* causing build under AST_DEVMODE to fail.
*
* The following PACKAGE_* macros are one place.
* Also netsnmp has an improper check for HAVE_DMALLOC_H, using
* #if HAVE_DMALLOC_H instead of #ifdef HAVE_DMALLOC_H
* As a countermeasure we define it to 0, however this will fail
* when the proper check is implemented.
*/
No
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
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r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines
In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.
While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generate loadable and embedded module lists.
Individual Makefiles now are a lot simpler, possibly as simple as this:
-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
MODULE_PREFIX=cdr_
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.
The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).
With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. When moh is started, we search first in memory to find the class. If we do not
find it in memory, we search realtime instead.
2. When moh is restarted (as in, it had been started on this particular channel, stopped,
and now we're starting it again), if using the "files" mode, then realtime will always
be rechecked. If you are using other modes, however, we will simply reattach to the external
running process which was playing moh earlier in the call. This is a necessary compromise so that
we don't end up with too many background processes.
3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory list. This has the advantage
of not requiring database lookups each time moh is started, but it has the disadvantage of not
truly being realtime.
I have tested this for functionality, and it passes. I also tested this under valgrind and there
are no memory problems reported under typical use.
Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker!
(closes issue #11196, reported and patched by sergee)
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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through ast_mutex primitives.
To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.
There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parked extension is provided as an argument.
(closes issue #10803)
Reported by: outtolunc
Patches:
res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237)
- modified by me to work a bit differently ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11237)
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r89248 | qwell | 2007-11-13 13:47:45 -0600 (Tue, 13 Nov 2007) | 7 lines
Revert change from revision 67064.
It is documented behavior that if a parking extension already exists while using PARKINGEXTEN,
dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you
must keep this in mind, and handle the failure yourself.
Issue 11237, reported by jon.
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Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
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r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line
In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
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the device that made the transfer instead. This makes for much smoother calling back
when queues are involved.
(closes issue #11155, reported by IPetrov)
Tremendous thanks to Russell for pulling me out of my block I was having on this one
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r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines
If someone were to delete the files used by an existing MOH class, and then
issue a reload, further use of that class could result in a crash due to
dividing by zero. This set of changes fixes up some places to prevent this
from happening.
(closes issue #10948)
Reported by: jcomellas
Patches:
res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282)
Additional changes added by me.
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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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r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 line
Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check.
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r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 line
closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config,
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Also fixes a few cli messages and some minor formatting.
(closes issue #11001)
Reported by: seanbright
Patches:
newcli.1.patch uploaded by seanbright (license 71)
newcli.2.patch uploaded by seanbright (license 71)
newcli.4.patch uploaded by seanbright (license 71)
newcli.5.patch uploaded by seanbright (license 71)
newcli.6.patch uploaded by seanbright (license 71)
newcli.7.patch uploaded by seanbright (license 71)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It allows you to configure a prefix for auto-monitor recordings.
(closes issue #6353)
Reported by: ivanfm
Patches:
asterisk_automon_v4.patch uploaded by ivanfm (original patch)
- updated patch:
6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Closes issue #10913, reported by tootai, who graciously granted us access
to his Asterisk server, thanks! Daniel, feel free to reopen the bug in
case you can reproduce this on 1.4.
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r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) | 5 lines
Presence packets from a client who's connected with our Jabber ID are
valid, therefore, those clients must be considered as buddies. The resource
string helps us make the distinction between clients.
Closes issue #10707, reported by yusufmotiwala.
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r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) | 5 lines
Prevent Asterisk from crashing when receiving a presence packet
without resource from a buddy that is known to have a resource list.
Revert a change I previously made, where Asterisk could point to a
freed memory location.
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r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1 line
closes issue #10834 ; where a null input to a switch statement results in a hangup; since switch is implemented with extensions, and the default case is implemented with a '.', and the '.' matches 1 or more remaining characters, the case where 0 characters exist isn't matched, and the extension isn't matched, and the goto fails, and a hangup occurs. Now, when a default case is generated, it also generates a single fixed extension that will match a null input. That extension just does a goto to the default extension for that switch. I played with an alternate solution, where I just tack an extra char onto all the patterns and the goto, but not the default case's pattern. Then even a null input will still have at least one char in it. But it made me nervous, having that extra char in , even if that's a pretty secret and low-level issue.
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r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) | 5 lines
Add another sanity check in the AGI read loop. We really don't care about
EAGAIN unless we didn't read an entire line. If there is a newline at the
end if the read buffer, break, because we got the whole thing.
(reported and patched by bmd)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84237 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)
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r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct 2007) | 6 lines
Fix randomness. save_pos was being set to 0 initially instead of -1, causing it to jump to position 0 when moh started.
(closes issue #10859)
Reported by: jamesgolovich
Patches:
asterisk-mohpos2.diff.txt uploaded by jamesgolovich (license 176)
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r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1 line
This issue sort of closes 10786; All config files support #include with globbing (you know, *,[chars],?,{list,list},etc), so I've updated the AEL system to support this also.
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r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1 line
This closes issue #10788 -- The exact same fixes are made here for the first arg in the for(arg1; arg2; arg3) {} statement, as were done for the 3rd arg. It can now be an assignment that will embedded in a Set() app, or a macro call, or an app call.
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r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1 line
This closes issue #10788 -- the 3rd arg in the for statement is now wrapped in Set() only if there's an '=' in that string. Otherwise, if it begins with '&', then a Macro call is generated; otherwise it is made into an app call. A bit more accomodating, keeps the new guys happy, and the guys with ael-1 code should be happy, too
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | 11 lines
Add a new patch to handle interrupting the fgets() call when using FastAGI.
This version of the patch maintains the original behavior of the code when
not using FastAGI.
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24)
Slight mods by me
Tested by: juggie, festr
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r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) | 5 lines
Handle the case where there are multiple dynamic features with the same digit
mapping, but won't always match the activated on/by access controls. In that
case, the code needs to keep trying features for a match.
(reported by Atis on the asterisk-dev list, patched by me)
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r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) | 9 lines
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-2.patch uploaded by juggie (license 24)
Tested by: juggie
When using fastagi, fgets() can return before a full line is read. Add explicit
handling for the case where it gets interrupted.
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r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines
Fix an issue that can occur when you do an attended transfer to parking. If
you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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Reported by: ornati
Make sure the called channel during the attended transfer process becomes associated with the calling channel so that the ast_waitfor_* call works properly under epoll.
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r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27 Aug 2007) | 11 lines
(closes issue #10419)
Reported by: mustardman
Patches:
asterisk-mohposition.diff.txt uploaded by jamesgolovich (license 176)
This patch fixes a few problems with music on hold.
* Fix issues with starting at the beginning of a file when it shouldn't.
* Fix the inuse counter to be decremented even if the class had not been
set to be deleted when not in use anymore
* Don't arbitrarily limit the number of MOH files to 255
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r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | 5 lines
When executing a dynamic feature, don't look it up a second time by digit pattern
after we already looked it up by name. This causes broken behavior if there is
more than one feature defined with the same digit pattern.
(closes issue #10539, reported by bungalow, patch by me)
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r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | 5 lines
Juggie in #asterisk-dev was reporting problems where fgets would return
without reading the whole line when using fastagi. When this happens,
errno was set to EINTR or EAGAIN. This patch accounts for the possibility
and lets fgets continue in that case.
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r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16 Aug 2007) | 14 lines
This patch fixes a bug where reloading the module with "module reload" did not
delete classes from memory that were no longer in the config. This patch fixes
that problem as well as another one. Previously, if you reloaded MOH using the
"moh reload" CLI command, which behaved differently than "module reload ...",
MOH had to be stopped on every channel and started again immediately. However,
there was no way to tell what class was being used, so they would all fall back
to the default class.
(closes issue #10139)
Reported by: blitzrage
Patches:
asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich
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r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007) | 21 lines
A fix for two critical problems detected while working with Daniel
McKeehan in issue #10184.
Upon priority change, the resource list is not NULL terminated when
moving an item to the end of the list. This makes Asterisk endlessy
loop whenever it needs to read the list. Jids with different resource and
priority values, like in Gmail's and GoogleTalk's jabber clients put
that problem in evidence.
Upon reception of a 'from' attribute with an empty resource string,
Asterisk crashes when trying to access the found->cap pointer if the
resource list for the given buddy is not empty. This situation is
perfectly valid and must be handled. The Gizmoproject's jabber client
put that problem in evidence.
Also added a few comments in the code as well as a handle for the
capabilities from Gmail's jabber client, which are stored in a caps:c tag
rather than the usual c tag.
Closes issue #10184.
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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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* Use a thread local ast_str for building the string that will be written out
to the console for debug, and to the FD for the AGI itself, instead of allocating
a buffer on the heap every time the function is called.
* Use the information contained within the ast_str to determine how many bytes
need to be written instead of calling strlen().
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r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines
(closes issue #10279)
Reported by: seanbright
Patches:
res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)
Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.
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r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines
(closes issue #10327)
Reported by: kkiely
Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place.
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- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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This merges the trunk only part of the patches from this issue. In 1.4, res_agi
will issue a warning if you try to use DeadAGI on a channel that is not hung up.
Now, in trunk, it just plain won't let you do it.
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r75067 | russell | 2007-07-13 15:10:40 -0500 (Fri, 13 Jul 2007) | 14 lines
Merged revisions 75059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 Jul 2007) | 6 lines
Ensure that adding a user to the list of users of a specific music on hold
class is not done at the same time as any of the other operations on this list
to prevent list corruption. Using the global moh_data lock for this is not
ideal, but it is what is used to protect these lists everywhere else in the
module, and I am only changing what is necessary to fix the bug.
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Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
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r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09 Jul 2007) | 9 lines
(closes issue #10123)
Reported by: blitzrage
Patches submitted by: juggie, qwell, me
Tested by: blitzrage
When trying to find a music on hold class to use, try all of the options,
instead of only the first one that is set. Also, change the MusicOnHold
applications to not hang up on the channel when a class can not be found.
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function to explicitly check for the int return value. Also, make a few
other minor changes such as removing a variable.
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r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) | 5 lines
Pass HOLD and UNHOLD frames to the other channel when they are returned from a
native bridge function. This fixes a problem where when two zap channels are
natively bridged and one does a flash hook, the other channel did not receive
music on hold. (Reported to me directly by Doug Bailey at Digium)
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for extracting application, function, manager, and agi documentation is the wrong
one to take. The most severe problem is that the output depends on which modules
are loaded as well as compile time options, which both determine which parts are
available.
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Terry Wilson
Tilghman Lesher
Kevin P. Fleming
Joshua Colp
Steve Murphy
Russell Bryant
Olle Johansson
Jason Parker
Philippe Sultan
Mark Michelson
Luigi Rizzo