The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(same patch as before, I just split this part out)
(close issue #12326)
Reported by: travishein
Patches:
app_voicemail_code_documentation.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem was that when the refcount on the queue hit 0, the destructor was
called, and inside the destructor, another function was called which would increase
the refcount back to 1 again and then decrease it again back to 0 for every member
in the queue. This meant that the destructor was being recursively called, leading
to a double free of the queue. This is now fixed by making sure to unlink the
queue from the queues container prior to the final unref of the queue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines
These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.
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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines
Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.
(closes issue #10487)
Reported by: ewilhelmsen
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1) Number of digits to enter can now be configured
2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue #7151)
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r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines
This patch makes it so that all queue member status changes are handled through device state
code. This removes several problems people were seeing where their queue members would get into
an "unknown" state. Huge props go to atis on this one since he was the one who found the code
section that was causing the problem and proposed the solution. I just wrote what he suggested :)
(closes issue #12127)
Reported by: atis
Patches:
12127v3.patch uploaded by putnopvut (license 60)
Tested by: atis, jvandal
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.
(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 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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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines
Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup(). The concept is exactly like the
fixup callback that is used in the channel technology interface. This callback
gets called when the owning channel changes due to a masquerade. Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.
(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)
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r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines
Fix a couple uses of sprintf. The second one could actually cause an overflow
of a stack buffer. It's not a security issue though, it only depends on your
configuration.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ctooley
Patches:
eivr_tcp_generic.patch uploaded by jpeeler (license 325)
This change adds the ability to communicate over a TCP socket instead of forking a child process.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines
(closes issue #12187, reported by atis, fixed by me after some brainstorming
on the issue with mmichelson)
- Update copyright info on app_chanspy.
- Fix a race condition that caused app_chanspy to crash. The issue was that
the chanspy datastore magic that was used to ensure that spyee channels did
not disappear out from under the code did not completely solve the problem.
It was actually possible for chanspy to acquire a channel reference out of
its datastore to a channel that was in the middle of being destroyed. That
was because datastore destruction in ast_channel_free() was done near the
end. So, this left the code in app_chanspy accessing a channel that was
partially, or completely invalid because it was in the process of being free'd
by another thread. The following sort of shows the code path where the race
occurred:
=============================================================================
Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
--------------------------------------||-------------------------------------
ast_channel_free() ||
- remove channel from channel list ||
- lock/unlock the channel to ensure ||
that no references retrieved from ||
the channel list exist. ||
--------------------------------------||-------------------------------------
|| channel_spy()
- destroy some channel data || - Lock chanspy datastore
|| - Retrieve reference to channel
|| - lock channel
|| - Unlock chanspy datastore
--------------------------------------||-------------------------------------
- destroy channel datastores ||
- call chanspy datastore d'tor ||
which NULL's out the ds' || - Operate on the channel ...
reference to the channel ||
||
- free the channel ||
||
|| - unlock the channel
--------------------------------------||-------------------------------------
=============================================================================
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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
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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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those XXX comments from the code.
The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are
not set, so there's no need to explicitly check them again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was the way that locks are referenced, since the old 1.2 names were still used
in the comments.
(closes issue #11997)
Reported by: snuffy
Patches:
bug_11997_queue_doxy.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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?
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r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb 2008) | 6 lines
When using autofill, members who are in use should be counted towards the
number of available members to call if ringinuse is set to yes.
Thanks to jmls who brought this issue up on IRC
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008) | 10 lines
This patch fixes some pretty significant problems with how app_chanspy handles
pointers to channels that are being spied upon. It was very likely that a
crash would occur if the channel being spied upon hung up. This was because
the current ast_channel handling _requires_ that the object is locked or else
it could disappear at any time (except in the owning channel thread). So, this
patch uses some channel datastore magic on the spied upon channel to be able to
detect if and when the channel goes away.
(closes issue #11877)
(patch written by me, but thanks to kpfleming for the idea, and to file for review)
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r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb 2008) | 8 lines
Clear up confusion when viewing the QUEUE_WAITING_COUNT of a
"dead" realtime queue. Since from the user's perspective, the queue
does exist, we shouldn't tell them we couldn't find the queue. Instead
since it is a dead queue, report a 0 waiting count
This issue was brought up on IRC by jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #11553)
Reported by: johan
Patches:
UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
CHANGES.channelredirect.patch uploaded by johan (license 334)
app_channelredirect-20080219.patch uploaded by johan (license 334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New feature: Add the 'e' option, which takes as an argument a list of
interfaces separated by colons. This way, you will only be able to spy
on this limited list of interfaces.
Bug fix: change some pointer checks to ast_strlen_zero so that spying
would work properly even if no channel was specified as the first argument
to chanspy.
(closes issue #10072)
Reported by: xmarksthespot
Patches:
bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by xmarksthespot (license 16)
Tested by: xmarksthespot, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
not be loaded from realtime queues. This commit fixes that.
Thanks to jmls for pointing this problem out to me on IRC.
This also contains some changes to S_OR where it should be used. Thanks to Qwell for pointing
these out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is done in a backward compat way.
If the "default" key for ffwd/rew is used for another option (such as stop), the "default" is removed.
(closes issue #11754)
Reported by: johan
Patches:
app_controlplayback.c.option3.patch uploaded by johan (license 334)
Tested by: johan, qwell
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r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan 2008) | 5 lines
Fix a logic error with regards to autofill. Prior to this change, it was possible
for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting
to call a member. This change fixes this.
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r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | 8 lines
ChanSpy issues a beep when it starts at the beginning of a list of channels to
potentially spy on. However, if there were no matching channels, it would beep
at you over and over, which is pretty annoying. Now, it will only beep once in
the case that there are no channels to spy on, but it will still beep again once
it reaches the beginning of the channel list again.
(closes issue #11738, patched by me)
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r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) | 8 lines
When we reset the password via an external command, we should also reset the
password stored in the in-memory list, too (otherwise it doesn't really take
effect).
(closes issue #11809)
Reported by: davetroy
Patches:
fix_externpass.diff uploaded by davetroy (license 384)
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so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:
"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.
1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."
This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.
Huge thanks to dimas for this wonderful submission.
(closes issue #11744)
Reported by: dimas
Patches:
dir3.patch uploaded by dimas (license 88)
Tested by: putnopvut, dimas
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r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines
Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.
Thanks to John Bigelow for pointing this error out to me.
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option tells JACK not to start jackd automatically if it is not already
running. Otherwise, the default is that jackd will get started for you if
it isn't running already.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
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r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines
Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely
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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)
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r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan 2008) | 5 lines
Part 1 of N of adding doxygen comments to app_queue. I picked some of the most common functions
used (which also happen to be some the biggest/ugliest functions too) to document first. I'm pretty
new to doxygen so criticism is welcome.
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that context to be entered as a new extension during the playback of a
voicemail greeting.
Patch inspired by bluecrow76, by tilghman.
(Closes issue #7063)
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will monitor this second device's state for the member, even though it actually calls the first
interface. This ability has been added for statically defined queue members, realtime queue members,
and dynamic queue members added through the CLI, dialplan, or manager.
(closes issue #11603, reported by acidv)
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r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines
Making some changes designed to not allow for a corrupted mailstream for a vm_state.
1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs.
2. Make sure to always grab the persistent vm_state when mailstream access is necessary.
3. Correct an incorrect return value in the init_mailstream function.
(closes issue #11304, reported by dwhite)
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go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.
(closes issue #11665, reported by yehavi)
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r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines
We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that
multiple members can have the same name, since the variable was not reset on each iteration of the loop.
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r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines
A change to improve the accuracy of queue logging in the case where a member does not
answer during the specified timeout period. Prior to this change, there was a small chance
that the member name recorded in this case would be blank. Also prior to this change, if using
the ringall strategy, if no one answered the call during the specified timeout, the member name
listed in the queue log would randomly be one of the members that was rung.
(closes issue #11498, reported and tested by hloubser, patched by me)
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of the queue_exec function by reversing the logic of an if statement. This change makes the function
comply better with the coding guidelines. Since this change is purely a cosmetic change to the code, I am
only committing the change to trunk.
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The channel name is printed in verbose messages
maximumWordLength option added.
Duration of words that do not meet the minimum word duration will be logged
The duration of pre-greeting silence will be logged
Only consider us in the greeting if we actually detected a valid word duration.
(closes issue #11650, reported and patched by davevg)
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r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec 2007) | 8 lines
I found a bug while browsing the queue code and managed to reproduce it in a small setup.
If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to
make app_queue think that all members at that penalty level were unavailable and cause the members at the
next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members
at a given penalty level are unreachable.
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Since the dtable in base_encode always gets populated with the same values every time and never
changes, make it static and const and only initialize it once. Also, there's no reason to
define BASEMAXINLINE twice, so remove the redundant #define.
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r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec 2007) | 5 lines
The mail_copy c-client function does not expect a full imap mailbox string, just the name of the mailbox.
(closes issue #11419, reported and patched by jaroth, with additional patchwork from me)
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queue members. This allows for the change in penalty levels to be executed at
the most logical time frame.
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the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending
on the amount of time passed. The purpose is to allow the call to open up to more (or maybe
just different) members without the caller's losing his place in the queue. See
configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample
for how to associate a rule with a queue.
Along with the functional changes, new CLI and manager commands exist to show the rules defined and
there is an additional CLI command to reload the queue rules.
Future enhancements that may be made: support for realtime queue rules and support for dynamically adding
a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write
this myself very soon).
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IMAP storage. The reason is that c-client has its own definitions for LOG_WARNING
and LOG_DEBUG, so we need to be sure to include asterisk's definitions last so that
we use the proper values in app_voicemail.
(closes issue #11437, reported by blitzrage, patch suggested by blitzrage)
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2. Fix an error when checking the CLI command for setting a member's penalty.
3. Fix a logging error if the incorrect parameter was the queue name or interface.
(closes issue #11544, reported and patched by Laureano)
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r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines
Issue 11574: Add dependencies on res_monitor and res_features.
I wonder if Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends on it.
Reported by: caio1982
(closes issue #11574)
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r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec 2007) | 6 lines
We need to create the directory for a voicemail user even if they are using IMAP storage
since greetings are stored in the filesystem.
(closes issue #11388, reported by spditner, patch by me inspired by a patch by spditner)
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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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r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec 2007) | 10 lines
Fixing autofill to be more accurate. Specifically, if calls ahead of the current
caller were ringing members (but not yet bridged) there could be available members
and waiting callers who would not get matched up. The member availability checker
was correctly determining the number of available members in this scenario, but
the queue itself did not parallelly reflect this status on the pending calls. This
commit corrects the issue.
(closes issue #11459, reported by equissoftware, patched by me)
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r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines
If there are no members in a queue, then the loop where the datastore for detecting
duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means
that when we try to free it, there's a crash. This stops that crash from occurring.
(closes issue #11499, reported by slavon, patched by eliel)
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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.
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r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines
* Add channel locking around datastore operations that expect the channel
to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Remove the dialed variable as it isn't needed.
* Restructure some code for clarity and coding guidelines stuff
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r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | 7 lines
* Add channel locking around datastore operations that expect the channel
to be locked.
* Document why we don't record Local channels in the dialed interfaces list.
* Handle memory allocation failure.
* Remove the dialed variable, as it wasn't actually needed.
* Tweak some formatting to conform to coding guidelines.
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r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | 7 lines
Fix in an issue in the call forwarding handling code that was causing crashes
on every call into a queue. I'm not entirely sure about the logic in this part
of the code, so I want to look at it some more tomorrow. However, this makes
it safe and keeps it from crashing.
(closes issue #11486, reported by adamg, patched by me)
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- Event Dial has new headers, to comply with other events
- Source -> Channel Channel name (caller)
- SrcUniqueID -> UniqueID Uniqueid
(new) -> Dialstring Dialstring in app data
(moremanager)
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r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines
The 'G' option for Dial() did not properly handle the case where only a label was
provided. This was due to the fact that the answering channel did not have an extension
set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto
on the answering channel since it is a wasteful call. The answering channel and the calling
channel are both directed to the same extension and context, just different priorities, so
we can just copy the values from the calling channel to the answering channel and increment
the answering channel's priority.
(closes issue #11382, reported by jon, patch by me with correction by jon)
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does not modify the contents of the "mailbox" string. In other words, I'm changing
the imap_retrieve_file function to take a const char* as the third argument so that I
don't need to cast const char*'s as char*'s to suppress compiler warnings.
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | 6 lines
The other day when I went through making changes as a result of the ao2_link()
change, I added some code to set pointers to NULL after they were unreferenced.
This pointed out that in this place, the object was unreferenced before the
code was done using it. So, move the unref down a little bit.
(crash reported by jmls on IRC)
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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)
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r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
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r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov 2007) | 6 lines
This patch handles the case where a queue member with a negative penalty is added
via the manager. If a negative value is submitted for a member penalty, we set it to 0.
(closes issue #11411, reported and patched by Laureano)
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the diff to trunk.
This just removes some checks on the return value of alloca(), as behavior
is undefined if it runs out of stack space, and we don't check it anywhere else.
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r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines
Two changes with regards to the 'eventwhencalled' option of queues.conf
1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to
'vars' or 'yes' did exactly the same thing. Thus the sign change of the
ast_true call.
2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting
in bizarre output for the channel variables. This patch remedies this.
(related to issue #11385, however I'm not sure if this will actually be enough to close it)
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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines
After issuing a "say load new", if a caller hangs up during the middle of playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.
(closes issue #11345, reported and patched by IgorG)
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- 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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r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines
Revert vmu->email back to an empty string if it was empty when imap_store_file
was called. This prevents sending a duplicate e-mail.
(closes issue #11204, reported by spditner, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines
Currently, zero-length voicemail messages cause a hangup in VoicemailMain.
This change fixes the problem, with a multi-faceted approach. First, we
do our best to avoid these messages from being created in the first place,
and second, if that fails, we detect when the voicemail message is
zero-length and avoid exiting at that point.
Reported by: dtyoo
Patch by: gkloepfer,tilghman
(Closes issue #11083)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Mark Michelson
Tilghman Lesher
Jason Parker
Joshua Colp
Steve Murphy
Russell Bryant
Terry Wilson
Jeff Peeler
Kevin P. Fleming
Olle Johansson
Michiel van Baak
Luigi Rizzo
Dwayne M. Hubbard
Doug Bailey