(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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
queue members. This allows for the change in penalty levels to be executed at
the most logical time frame.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92203 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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Mark Michelson
Michiel van Baak
Tilghman Lesher
Russell Bryant
Jason Parker
Olle Johansson
Kevin P. Fleming
Terry Wilson
Luigi Rizzo
Dwayne M. Hubbard
Doug Bailey