A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Check doxygen for more details
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115268 f38db490-d61c-443f-a65b-d21fe96a405b
........
r115257 | bbryant | 2008-05-02 15:25:42 -0500 (Fri, 02 May 2008) | 2 lines
Add new "pri show version" command to show the libpri version for support reasons.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115258 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines
Clarify a comment that was, well, just wrong. It turns out that
ignoring the way that macros expand. Instead, I have clarified in the
comment why the macro will work even if the scheduler id for the
task to be deleted changes during the execution of the macro.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115197 f38db490-d61c-443f-a65b-d21fe96a405b
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115021 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) | 6 lines
'#' is another reserved character for URIs that also needs to be escaped.
(closes issue #10543)
Reported by: blitzrage
Patches:
20080418__bug10543.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115018 f38db490-d61c-443f-a65b-d21fe96a405b
for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114931 f38db490-d61c-443f-a65b-d21fe96a405b
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified.
(closes issue #12536)
Reported by: bjm
Patches:
12536-dringXrange.diff uploaded by qwell (license 4)
Tested by: bjm
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114922 f38db490-d61c-443f-a65b-d21fe96a405b
to announce-position, "limit" and "more," as well as a new option,
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.
(closes issue #10991)
Reported by: slavon
Patches:
app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114906 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@114904 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114892 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114884 f38db490-d61c-443f-a65b-d21fe96a405b
a colon-delimited list of spygroups to be specified when calling the ChanSpy application
with the 'g' option. Prior to this, you could only specify a single group when using the
'g' option.
I also have upped the maximum number of spygroups to 128 and added a #define so that this
can be easily increased or decreased later.
(closes issue #12497)
Reported by: jsmith
Patches:
app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license 15)
Tested by: atis, jvandal
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114857 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines
Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.
(closes issue #12549)
Reported by: darren1713
Patches:
app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
(with modifications from me)
Tested by: putnopvut
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114849 f38db490-d61c-443f-a65b-d21fe96a405b
functions in app_directory can be removed since the ODBC-specific lookups are accomplished
within app_voicemail. This change greatly reduces the amount of lines in app_directory that
were solely for the purpose of looking up a name when ODBC_STORAGE is specified for voicemail.
This commit also makes the name-saying interruptable via DTMF.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114834 f38db490-d61c-443f-a65b-d21fe96a405b
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114813 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114709 f38db490-d61c-443f-a65b-d21fe96a405b
(closes issue #12214)
Reported by: DEA
Patches:
chan_skinny-vm-events-v3.txt uploaded by DEA (license 3)
Tested by: DEA and me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114700 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines
When we don't explicitly pass a path to the --with-tds configure option, we
may end up finding tds.h in /usr/local/include instead of /usr/include. If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114696 f38db490-d61c-443f-a65b-d21fe96a405b
barge on the call. It is like the existing whisper option, except that
it allows the spy to talk to both sides of the conversation on which
he is spying.
This feature has existed in Switchvox, and this merges the functionality
into Asterisk.
(AST-32)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114678 f38db490-d61c-443f-a65b-d21fe96a405b
dhubbard