Commit Graph

3668 Commits

Author SHA1 Message Date
David Ruggles
4c49e70ec5 Add send DTMF feature to ExternalIVR app
Implemented a new command 'D' that allows client
IVRs to send DTMF digits to the channel.

(closes issue #16615)
Reported by: thedavidfactor

Review: https://reviewboard.asterisk.org/r/465/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-22 16:20:43 +00:00
Tilghman Lesher
873989db91 Enable SendText to send strings in encoded format.
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19 22:41:36 +00:00
David Ruggles
174cd3c65c Add notification of interrupted file
Add file information to data element of T event so
the file information is sent to the client when it is
interrupted. Previously only notification of pending
files that were dropped was sent

(closes issue #16147)
Reported by: thedavidfactor
Tested by: thedavidfactor

Review: https://reviewboard.asterisk.org/r/449/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-18 17:41:44 +00:00
David Vossel
8d8800072e fixes spelling error. s/memeber/member
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-18 15:52:55 +00:00
Tilghman Lesher
e8a6d2995e Add pickup event to AMI. Also, fix AMI documentation.
(closes issue #16431)
 Reported by: syspert
 Patches: 
       20100112__issue16431.diff.txt uploaded by tilghman (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15 21:04:34 +00:00
Tilghman Lesher
f94e723a27 Make sure that the limit is N, not N - 1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15 20:58:19 +00:00
Tilghman Lesher
6bc1fc7240 Merged revisions 240414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 Jan 2010) | 15 lines
  
  Disallow leaving more than maxmsg voicemails.
  This is a possibility because our previous method assumed that no messages are
  left in parallel, which is not a safe assumption.  Due to the vmu structure
  duplication, it was necessary to track in-process messages via a separate
  structure.  If at some point, we switch vmu to an ao2-reference-counted
  structure, which would eliminate the prior noted duplication of structures,
  then we could incorporate this new in-process structure directly into vmu.
  (closes issue #16271)
   Reported by: sohosys
   Patches: 
         20100108__issue16271.diff.txt uploaded by tilghman (license 14)
         20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14)
         20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14)
   Tested by: jsutton
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15 20:54:24 +00:00
Sean Bright
e612d87695 Convert a few places to use ast_calloc_with_stringfields where applicable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15 18:21:50 +00:00
David Vossel
03529837cc add silence gen to wait apps
asterisk.conf's 'transmit_silence' option existed before
this patch, but was limited to only generating silence
while recording and sending DTMF.  Now enabling the
transmit_silence option generates silence during wait
times as well.

To achieve this, ast_safe_sleep has been modified to
generate silence anytime no other generators are present
and transmit_silence is enabled.  Wait apps not using
ast_safe_sleep now generate silence when transmit_silence
is enabled as well.

(closes issue #16524)
Reported by: kobaz

(closes issue #16523)
Reported by: kobaz
Tested by: dvossel

Review: https://reviewboard.asterisk.org/r/456/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13 16:31:14 +00:00
TransNexus OSP Development
912d4da476 Updated XML doc for OSP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13 07:00:13 +00:00
David Vossel
0a6c0ee1f7 cli 'queue show' formatting fix. queue name was truncated over 12 characters
(closes issue #16078)
Reported by: RoadKill
Patches:
      quequename_limit.patch uploaded by ppyy (license 906)
Tested by: dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07 18:58:23 +00:00
Jeff Peeler
c6e038ba16 Fix misreverting from 177158.
(closes issue #15725)
Reported by: shanermn
Patches: 
      v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-06 20:37:18 +00:00
Russell Bryant
5d7b80248b Merged revisions 238009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines
  
  Resolve a crash due to an ast_frame not being fully initialized.
  
  (closes issue #16531)
  Reported by: john8675309
  
  (closes SWP-615)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-06 15:19:10 +00:00
David Vossel
bfae8dca78 fixes holdtime playback issue in app_queue
When reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123 seconds"
rather than "2 minutes 3 seconds".

Also, the "minute" sound file is missing, so for the moment until
that file can be created the "minutes" file is used instead.

(closes issue #16168)
Reported by: nickilo
Patches:
      patch-unified-trunk-rev-222176 uploaded by nickilo (license )
Tested by: nickilo, wonderg



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05 23:08:50 +00:00
Mark Michelson
2fa64b3ad4 Mismerged a bit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05 20:56:50 +00:00
Mark Michelson
c9d1ffcae8 Add a missing part of the connected line work into trunk.
Part of the work done for connected line was to add an optional
argument to the 'f' option to allow for the connected party information
of the outgoing channel to be set to the argument provided. This was
overlooked during the merge of the work to trunk and is being added
back now. The CHANGES file has also been updated to note this change.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05 18:46:19 +00:00
Michiel van Baak
0c62434201 Make CLI command 'mixmonitor start|stop <channel> work again.
(closes issue #16534)
Reported by: jlaguilar
Fix as suggested by jlaguilar in the bugreport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05 16:08:12 +00:00
David Vossel
688e1bbac6 app_queue segfaults if realtime field uniqueid is NULL
(closes issue #16385)
Reported by: haakon
Patches:
      app_queue.c.patch uploaded by haakon (license 880)
      app_queue.c.patch_v2 uploaded by dvossel (license 671)
Tested by: haakon



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04 16:39:11 +00:00
TransNexus OSP Development
fb4870a74a 1. Added reporting operator names in AuthReq.
2. Added retrieving operator names from AuthRsp and exporting them.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04 03:38:29 +00:00
Jason Parker
f93071483f Add app_voicemail and say.c support for Vietnamese.
Also add an XXX comment that I'm baffled nobody has ever complained about.  We
say "first message", and then we go into language-specific stuff where we
proceed to say..."first message".

(closes issue #15053)
Reported by: dinhtrung
Patches:
      vietnamese.ods uploaded by dinhtrung (license 776)
      app_voicemail.c.diff uploaded by dinhtrung (license 776)

(closes issue #15626)
Reported by: dinhtrung
Patches:
      say.c.diff uploaded by dinhtrung (license 776)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-30 22:30:21 +00:00
TransNexus OSP Development
28d16a3cb1 1. Updated for OSP Toolkit 3.6.0.
2. Added service type ported number query.
3. Formated code.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-29 10:59:55 +00:00
Tilghman Lesher
06dc97772e Use recommended option, not deprecated option.
(closes issue #16515)
 Reported by: ManChicken


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-28 17:37:46 +00:00
Sean Bright
82446789f3 Merged revisions 236509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
  
  Avoid a crash with large numbers of MeetMe conferences.
  
  Similar to changes made to Queue(), when we have large numbers of conferences in
  meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
  crash, so instead just use a single fixed buffer.
  
  (closes issue #16509)
  Reported by: Kashif Raza
  Patches:
        20091223_16509.patch uploaded by seanbright (license 71)
  Tested by: seanbright
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-28 12:44:58 +00:00
David Vossel
0a5d21e6c7 QUEUE_MEMBER(..., ready) counts only ready agents, not free agents wrapping up
The QUEUE_MEMBER dialplan function can return total members,
logged-in members and "free" members count. A member is counted
as "free" immediately after his call ends, even though its wrap-up
time, if specified in queues.conf, has not yet expired, and the
queue will not actually route a call to it.

This Patch introduces a new "ready" option that only counts
free agents no longer in the wrap up time period.

(closes issue #16240)
Reported by: kkm
Patches:
      appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888)
Tested by: kkm, dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23 19:14:05 +00:00
David Vossel
065fce7310 update CHANGES to reflect new 'R' app_queue option plus a minor optimization to the feature patch
(issue #16384)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23 18:45:54 +00:00
David Vossel
6892b103ab new parameter 'R' to the Queue application
The 'R' argument stops moh and indicates ringing once the agent is
ringing.  This allows the person in the queue to know their call
is potentially about to be answered.

(closes issue #16384)
Reported by: haakon
Patches:
      new_app_queue.c.patch uploaded by haakon (license 880)
Tested by: haakon, loloski, dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23 18:39:37 +00:00
Tilghman Lesher
169b74c313 AGI may be invoked from outside the dialplan
(closes issue #16510)
 Reported by: atis
 Patches: 
       20091223__issue16510.diff.txt uploaded by tilghman (license 14)
 Tested by: atis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23 18:25:27 +00:00
Tilghman Lesher
1e0306a04b Actually use tmp for something (brings trunk back into sync with 1.6 branches).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23 02:52:30 +00:00
Alec L Davis
7537d3c0cb app_dial optional parameter to option 'r' to allow play indication from indications.conf
(closes issue #14504)
  Reported by: alecdavis
  Tested by: alecdavis,jsmith
  Patch
	 app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-19 08:59:31 +00:00
Kevin P. Fleming
df1fc1f381 spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 14:35:46 +00:00
Alec L Davis
13c3260c92 Support option 'n', as applications like Playback, Background etc.
Suggested on asterisk-dev as trivial application change.
 
Reported by: alecdavis
Tested by: alecdavis


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 07:18:31 +00:00
Alec L Davis
90be4cf5ef fixes escape to extensions 'o' and 'a', for digits '0' and '*'
(closes issue #16437)
Reported by: alecdavis
Tested by: alecdavis
Patch
	extension_o_a_fix.diff.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 02:29:50 +00:00
Alec L Davis
19f8080654 ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.
(closes issue #16409)
  Reported by: alecdavis
  Tested by: alecdavis
  Patch
	bug_16409.diff.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 00:54:44 +00:00
Tilghman Lesher
89a1af1d38 Allow greetings-only mailboxes for Voicemail.
(closes issue #15132)
 Reported by: floletarmo
 Patches: 
       voicemail_changes.patch uploaded by floletarmo (license 784)
       (with some additional changes by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14 23:16:00 +00:00
Jason Parker
e52ee5c8e6 Allow tonelist as argument to ReadExten.
ReadExten already supported playing a tonezone from indications.conf.
It now has the ability to use a tonelist like 440+480/2000|0/4000

(closes issue #15185)
Reported by: jcovert
Patches:
      app_readexten.c.patch uploaded by jcovert (license 551)
Tested by: qwell
Patch modified by me, to maintain backwards compatibility.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14 21:32:03 +00:00
Jeff Peeler
2923086daf Merged revisions 234379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
  
  Fix talking detection status after conference user is muted.
  
  This patch ensures that when a conference user is muted that the accompanying
  AMI Meetme talking off event is sent. Also, the meetme list output is updated
  to show the muted user as unmonitored.
  
  (closes issue #16247)
  Reported by: dimas
  Patches: 
        v3-16247.patch uploaded by dimas (license 88)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-11 23:17:09 +00:00
Jeff Peeler
2414bc8005 Add audio announcement option to app_page
As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
  conference.
* Page has a new option 'A(x)' which will playback an announcement 
  simultaneously to all paged phones (and optionally excluding the caller's one 
  using the new option 'n') before the call is bridged.

To add the new option to meetme, the conference flag options had to be extended 
to 64 bits.

(closes issue #14365)
Reported by: dferrer
Patches:
      page_announce.patch uploaded by dferrer (license 525)
      modified by me

Review: https://reviewboard.asterisk.org/r/188/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-10 17:31:23 +00:00
David Ruggles
43ebe5a2ba Fix TCP Client interface
Fix a couple of very minor bugs that prevent the socket client from working. The wrong set of properties were used in one place and the size of the address variable isn't set if the host name is an ip address. Also includes a fix for a bug that was introduced previously.

(closes issue #16121)
Reported by: thedavidfactor
Tested by: thedavidfactor

Review: https://reviewboard.asterisk.org/r/439/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-07 19:48:14 +00:00
David Vossel
1c539e6982 .m3u support for Mp3Player app
(closes issue #14823)
Reported by: macli
Patches:
      app_mp3.diff1 uploaded by macli (license )
Tested by: macli, dvossel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 20:19:53 +00:00
David Vossel
63dafe98f6 changes penaltymemberslimit to use scanf for config value parsing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 18:55:21 +00:00
David Vossel
e21deabf02 new queue option, penaltymemberslimit, disregards penalty on too few queue members when enabled
(closes issue #14559)
Reported by: fiddur
Patches:
      trunk-199584-1.diff uploaded by fiddur (license 678)
Tested by: fiddur, dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 18:48:31 +00:00
David Vossel
f72b2a060d Merged revisions 233116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines
  
  document and rename strip_control() in app_voicemail
  
  (closes issue #16291)
  Reported by: wdoekes
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 17:22:31 +00:00
Tilghman Lesher
8599628e0b Add pagerdateformat, to allow shorter dates for SMS messages.
(closes issue #16263)
 Reported by: andrew
 Patches: 
       pagerdate.patch uploaded by andrew (license 240)
       (with a slight modification by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03 22:13:56 +00:00
Tilghman Lesher
d75ebf8afc Merged revisions 232820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines
  
  Deprecate "cz" in favor of "cs".
  Also, change the use of language codes so that language registers as a prefix,
  rather than an exact match.
  (closes issue #16272)
   Reported by: patrol-cz
   Patches: 
         20091203__issue16272.diff.txt uploaded by tilghman (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03 20:47:07 +00:00
TransNexus OSP Development
afee39cb4c Replaced two deprecated functions of OSP Toolkit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03 08:47:38 +00:00
TransNexus OSP Development
8c69320c87 Added custom info support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03 03:56:14 +00:00
Jeff Peeler
e5aa8cad9b Extend voicemail to allow IMAP folders to be specified per mailbox.
Previously only possible per context, new option called imapfolder.

(closes issue #14298)
Reported by: jablko
Patches: 
      patch-200906202 uploaded by jablko (license 675)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03 00:38:03 +00:00
David Ruggles
93afa4cc4f Prevent double closing of FDs by EIVR
This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications.
EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance
the second close would then close the FD now in use by AGI.

(closes issue #16305)
Reported by: diLLec
Tested by: thedavidfactor, diLLec

Review: https://reviewboard.asterisk.org/r/436/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 22:17:22 +00:00
Joshua Colp
a0b1c785c6 Add an option to Record which enables a mode where any DTMF digit will terminate recording.
(closes issue #15436)
Reported by: Vince
Patches:
      app_record.diff uploaded by Vince (license 823)
Tested by: dbrooks


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 18:35:47 +00:00
Joshua Colp
f050ba6b38 Merged revisions 232355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines
  
  Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
  
  (closes issue #16239)
  Reported by: CGMChris
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 17:06:54 +00:00