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Author SHA1 Message Date
mmichelson 67e82aa1ba Merged revisions 91292 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec 2007) | 3 lines

Reverting extra stuff I didn't mean to commit


........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91293 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05 22:57:57 +00:00
mmichelson eedde68808 Merged revisions 91273 via svnmerge from
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)


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91291 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05 22:55:49 +00:00
russell 0738ce5f9e Resolve compiler warnings.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91193 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05 17:44:59 +00:00
tilghman 29a6f4d8e6 Added multiple name listing. (Closes issue #10413)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05 16:25:52 +00:00
mmichelson b31b092cff Kevin suggested doing the reverse of my last commit, since imap_retrieve_file
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: http://svn.digium.com/svn/asterisk/trunk@90930 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 18:29:35 +00:00
mmichelson fda84d629d Suppress a compiler warning due to discarding a "const" qualifier
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90928 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 18:14:08 +00:00
mmichelson 48f394751c Wrong locking style got merged from 1.4 to trunk. My mistake.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90899 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 17:51:59 +00:00
qwell 807b019f34 Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 17:35:40 +00:00
mmichelson 4128bac7a9 Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 17:08:36 +00:00
oej 4df981b891 (closes issue #11431)
Reported by: Laureano
Patches: 
      app_queue.c.patch uploaded by Laureano (license 265)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90854 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04 15:16:03 +00:00
qwell 2ee1c89345 Merged revisions 90696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

(Closes issue #11383)
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r90696 | qwell | 2007-12-03 16:06:36 -0600 (Mon, 03 Dec 2007) | 4 lines

Make sure we always close the conference fd if we have an open one.

Issue 11383, reported by markmhy, patch by eliel.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90697 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03 22:07:57 +00:00
mmichelson 893a67e791 Replacing some calls to free() with ast_free().
(closes issue #11448, reported and patched by jaroth)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90670 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03 21:24:56 +00:00
file 344ae791b9 Remove the file descriptors from the main poll channel when the channel is hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end.
(closes issue #11441)
Reported by: blitzrage


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90508 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03 14:14:43 +00:00
russell 814e0a28cf Merged revisions 90470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90471 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-02 18:20:13 +00:00
mmichelson a7c0447e63 Adding support for the "automixmonitor" dial and queue options.
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: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30 21:19:57 +00:00
russell 7cfa10f05b Merged revisions 90348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90351 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30 19:34:47 +00:00
mmichelson 18943278d5 Merged revisions 90163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90164 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29 19:39:31 +00:00
file 9cebea59f4 Merged revisions 90101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines

Fix a few memory leaks.
(closes issue #11405)
Reported by: eliel
Patches:
      load_realtime.patch uploaded by eliel (license 64)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90102 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28 23:03:09 +00:00
russell 1d81780e8d Merge some little changes from team/russell/chan_refcount to help reduce
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.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89947 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28 00:49:55 +00:00
russell 45c611df49 Merged revisions 89844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines

Instead of depending on the return value of ast_true(), explicitly set the
eventwhencalled variable to 1.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89847 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 23:21:38 +00:00
mmichelson c6423f296f Merged revisions 89837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89838 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 23:11:12 +00:00
murf 5aff21b945 Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 06:47:08 +00:00
mmichelson bbc2a86e79 Merged revisions 89618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89619 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 23:11:29 +00:00
oej d33873fade - Mark "concise" as deprecated
- 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?


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 19:24:23 +00:00
file 78ee740bbf Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89602 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 18:11:31 +00:00
file c585cac92c Merged revisions 89587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines

Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
      mixmonitor.patch uploaded by reformed (license 330)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89589 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:23:28 +00:00
kpfleming 9af283654a Merged revisions 89586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines

when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89588 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:21:37 +00:00
mmichelson 43aebe686f Merged revisions 89580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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: http://svn.digium.com/svn/asterisk/trunk@89581 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 15:50:37 +00:00
file 8ad982f1ba Merged revisions 89571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 lines

When unloading app_meetme destroy any auto created contexts created by SLA.
(closes issue #11367)
Reported by: eliel

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89572 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 14:42:57 +00:00
file b8f8021a8a Don't crash if the 'o' option of ControlPlayback is used without any value.
(closes issue #11375)
Reported by: johan


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89570 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 14:31:32 +00:00
tilghman cdbe5cdfb1 Merged revisions 89540 via svnmerge from
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: http://svn.digium.com/svn/asterisk/trunk@89541 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24 06:24:46 +00:00
rizzo d34da1d386 more header removal
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89524 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 04:37:08 +00:00
rizzo e8a5f98fe8 shuffle a little bit the content of header files to reduce dependencies.
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: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 03:50:04 +00:00
rizzo 15e517d2ae remove some useless includes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89521 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 02:30:58 +00:00
rizzo 737b408d52 more removal of redundant headers
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89519 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 02:07:33 +00:00
rizzo c94efd7d1e remove redundant headers
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89518 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 01:39:06 +00:00
rizzo 8cd33321ef remove a number of #include <fcntl.h> which are either
useless or done elsewhere



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 01:03:02 +00:00
murf b6e2980dd6 closes issue #11285, where an unload of a module that creates a dialplan context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89513 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 23:54:12 +00:00
rizzo 150b2c22ef remove another set of redundant #include "asterisk/options.h"
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 23:24:55 +00:00
mmichelson c6d5070e7d Merged revisions 89495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov 2007) | 3 lines

Fix a small error I made in my previous commit


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89496 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 19:28:43 +00:00
mmichelson ceb701426d Merged revisions 89493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines

Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this
message would always report that there were 0 members available, even though that may not be true.



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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89494 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 19:27:22 +00:00
tilghman 21981c69ae Change Read to set READSTATUS as an indication of the result
Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89489 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 18:38:18 +00:00
russell 855aea6095 Merge changes from team/russell/sla_trunk_moh ...
* Added the ability to specify the music on hold class used to play into the
   conference when there is only one member and the M option is used.
* Added the ability to specify a music on hold class to play instead of ringing
   for the SLATrunk application.

(patched by me, and tested internally)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89470 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 00:21:38 +00:00
tilghman 1688610395 Make trunk build again
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89468 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20 23:29:33 +00:00
rizzo 89d8d78652 move asterisk/paths.h outside asterisk.h and into those files
who really need it.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20 23:16:15 +00:00
rizzo 03ef197f9e Fix building of modules under cygwin.
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: http://svn.digium.com/svn/asterisk/trunk@89454 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20 16:12:10 +00:00
rizzo 7f3cce8be2 more errno.h removal
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89432 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 21:12:08 +00:00
rizzo 9cf442d7f7 include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere.
Remove some of the instances.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 18:52:04 +00:00
file 9ba23154e5 Change warning messages (which are really debug messages) into debug messages.
(closes issue #11288)
Reported by: IgorG
Patches:
      saydebug-89394-1-trunk.patch uploaded by IgorG (license 20)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89410 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 14:03:30 +00:00
rizzo 36281f41ce another cygwin compatibility fix.
This one must be handled in a better way in configure, also for other
architectures



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89374 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-17 11:08:58 +00:00