[asterisk-commits] tilghman: branch group/codec_bits r116662 - in /team/group/codec_bits: ./ app...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 15 14:48:23 CDT 2008
Author: tilghman
Date: Thu May 15 14:48:22 2008
New Revision: 116662
URL: http://svn.digium.com/view/asterisk?view=rev&rev=116662
Log:
Merged revisions 115507,115509,115513,115515,115518-115519,115521,115523,115525,115535,115537,115546,115548,115552,115555,115558,115562,115566,115569,115580,115582,115584,115586,115588,115591,115593-115596,115598,115600,115669,115705,115737,115784,115813,115847,115850,115886,115939,115941,115945,116001,116039,116089,116138,116179,116222-116224,116229,116234,116237,116239-116240,116297-116298,116350,116353,116407,116410,116461,116467,116469,116471,116522,116557,116590,116592,116594,116631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r115507 | file | 2008-05-07 08:41:25 -0500 (Wed, 07 May 2008) | 4 lines
Remove redundant header getting.
(closes issue #12597)
Reported by: hooi
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r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 May 2008) | 6 lines
Update typos in description fields
(closes issue #12598)
Reported by: suretec
Patches:
asterisk_schema_changes.patch uploaded by suretec (license 70)
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r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines
Merged revisions 115512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines
Merged revisions 115511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines
Remove remnants of dlinkedlists. I didn't actually use them in the final version
of my IAX2 improvements.
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r115515 | russell | 2008-05-07 12:38:36 -0500 (Wed, 07 May 2008) | 2 lines
re-add dlinkedlists.h to trunk, oops!
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r115518 | russell | 2008-05-07 13:17:43 -0500 (Wed, 07 May 2008) | 12 lines
Blocked revisions 115517 via svnmerge
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r115517 | russell | 2008-05-07 13:17:19 -0500 (Wed, 07 May 2008) | 5 lines
Track peer references when stored in the sip_pvt struct as the peer related to
a qualify ping or a subscription. This fixes some realtime related crashes.
(closes issue #12588)
(closes issue #12555)
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r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines
Let chan_h323 build in dev mode
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r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) | 7 lines
Use the default that the log output claims will be used for the basedn
(closes issue #12599)
Reported by: suretec
Patches:
12599.patch uploaded by juggie (license 24)
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r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) | 6 lines
Only save a password if a username exists.
(closes issue #12600)
Reported By: suretec
Patch by me
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r115525 | tilghman | 2008-05-07 13:40:21 -0500 (Wed, 07 May 2008) | 2 lines
Don't free the object on destroy, as astobj2 takes care of that for you
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r115535 | tilghman | 2008-05-07 15:22:09 -0500 (Wed, 07 May 2008) | 2 lines
Advance to next sounds release
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r115537 | russell | 2008-05-07 16:11:33 -0500 (Wed, 07 May 2008) | 10 lines
Fix up a problem that was introduced into the scheduler when it was converted
to use doubly linked lists. The schedule() function had an optimization that
had it try to guess which direction would be better for the traversal to insert
the task into the scheduler queue. However, if the code chose the path where
it traversed the queue in reverse, and the result was that the task should be
at the head of the queue, then the code would actually put it at the tail,
instead.
(Problem found by bbryant, debugged and fixed by bbryant and me)
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r115546 | russell | 2008-05-08 09:41:12 -0500 (Thu, 08 May 2008) | 12 lines
Merged revisions 115545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) | 4 lines
Use the same method for executing Asterisk as the rest of the script.
(closes issue #12611)
Reported by: b_plessis
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r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line
Remove unused code as well as demote an error message to a debug message
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r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) | 12 lines
Merged revisions 115551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) | 4 lines
Don't use a channel before checking for channel allocation failure.
(closes issue #12609)
Reported by: edantie
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r115555 | russell | 2008-05-08 10:32:48 -0500 (Thu, 08 May 2008) | 11 lines
Merged revisions 115554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) | 3 lines
Don't exit the script if Asterisk is not running.
(closes issue #12611)
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r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) | 11 lines
Merged revisions 115557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) | 3 lines
remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well
(closes issue #9676)
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r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) | 11 lines
Merged revisions 115561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines
Don't give up on attempting an outbound registration if we receive a 408 Timeout.
(closes issue #12323)
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r115566 | russell | 2008-05-08 14:17:04 -0500 (Thu, 08 May 2008) | 41 lines
Merged revisions 115565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines
Merged revisions 115564 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines
Fix a race condition that bbryant just found while doing some IAX2 testing.
He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes,
however, the audio was extremely choppy. We looked at a packet trace and saw
a storm of INVAL and VNAK frames being sent from one box to another.
It turned out that what had happened was that one box tried to send a CONTROL
frame before the 3 way handshake had completed. So, that frame did not include
the destination call number, because it didn't have it yet. Part of our recent
work for security issues included an additional check to ensure that frames that
are supposed to include the destination call number have the correct one. This
caused the frame to be rejected with an INVAL. The frame would get retransmitted
for forever, rejected every time ...
This race condition exists in all versions that got the security changes,
in theory. However, it is really only likely that this would cause a problem in
Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing
all versions that could potentially be affected by the introduced race condition.
These changes are what bbryant and I came up with to fix the issue. Instead of
simply dropping control frames that get sent before the handshake is complete,
the code attempts to wait a little while, since in most cases, the handshake
will complete very quickly. If it doesn't complete after yielding for a little
while, then the frame gets dropped.
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r115569 | russell | 2008-05-08 14:20:35 -0500 (Thu, 08 May 2008) | 10 lines
Merged revisions 115568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines
Remove debug output.
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r115580 | file | 2008-05-09 11:36:58 -0500 (Fri, 09 May 2008) | 10 lines
Merged revisions 115579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines
Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform.
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r115582 | tilghman | 2008-05-09 12:28:06 -0500 (Fri, 09 May 2008) | 7 lines
Allow a password change to be validated by an external script.
(closes issue #12090)
Reported by: jaroth
Patches:
vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7)
20080509__bug12090.diff.txt uploaded by Corydon76 (license 14)
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r115584 | bbryant | 2008-05-09 14:54:45 -0500 (Fri, 09 May 2008) | 15 lines
The following patch adds new options and alters the default behavior of the ENUM* functions. The TXCIDNAME lookup function has also gotten a
new paramater. The new options for ENUM* functions include 'u', 's', 'i', and 'd' which return the full uri, trigger isn specific rewriting, look
for branches into an infrastructure enum tree, or do a direct dns lookup of a number respectively. The new paramater for TXCIDNAME adds a
zone-suffix argument for looking up caller id's in DNS that aren't e164.arpa.
This patch is based on the original code from otmar, modified by snuffy, and tested by jtodd, me, and others.
(closes issue #8089)
Reported by: otmar
Patches:
20080508_bug8089-1.diff
- original code by otmar (license 480),
- revised by snuffy (license 35)
Tested by: oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant
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r115586 | bbryant | 2008-05-09 15:05:50 -0500 (Fri, 09 May 2008) | 2 lines
Update CHANGES file for previous commit of ENUM and TXCIDNAME changes.
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r115588 | mmichelson | 2008-05-09 16:22:42 -0500 (Fri, 09 May 2008) | 19 lines
Adding support for "urgent" voicemail messages. Messages which are
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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r115591 | mmichelson | 2008-05-09 17:36:50 -0500 (Fri, 09 May 2008) | 3 lines
Remove a debug line
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r115593 | junky | 2008-05-09 22:04:25 -0500 (Fri, 09 May 2008) | 2 lines
since we unregister, that has not been properly registered, i standardized this.
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r115594 | junky | 2008-05-09 22:28:50 -0500 (Fri, 09 May 2008) | 3 lines
ameliorate load and unload to dont use DECLINED or FAILED, when theres no .conf involved.
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r115595 | junky | 2008-05-09 22:30:59 -0500 (Fri, 09 May 2008) | 3 lines
fix a sample since we now required , and not | for the arguments separator
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r115596 | tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines
Ensure that "calldate" is acceptable for a column name.
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r115598 | mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line
Open up audio channel when we get ACM on SS7 event
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r115600 | mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line
Add Zap MTP2 support to chan_zap
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r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) | 3 lines
A small change to fix iax2 native bridging.
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r115705 | qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line
Correctly document state interface for AddQueueMember. Discovered while looking at issue #12626.
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r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May 2008) | 15 lines
Merged revisions 115735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines
If a thread holds no locks, do not print any information on the thread when issuing
a core show locks command. This will help to de-clutter output somewhat.
Russell said it would be fine to place this improvement in the 1.4 branch, so that's
why it's going here too.
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r115784 | oej | 2008-05-12 13:39:09 -0500 (Mon, 12 May 2008) | 4 lines
Add support for playing an audio file for caller and callee at start and stop of monitoring (one-touch monitor).
Keep messages short, since the other party is waiting while one party hear the message...
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r115813 | tilghman | 2008-05-12 15:34:38 -0500 (Mon, 12 May 2008) | 8 lines
Add a script which installs every package needed for a Debian install of
Asterisk, and includes possible support (to be contributed) for various other
distributions.
(closes issue #10523)
Reported by: tzafrir
Patches:
install_prereq_2 uploaded by tzafrir (license 46)
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r115847 | russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines
Initialize the start time in smdi_msg_wait. Somehow this code got lost in trunk.
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r115850 | russell | 2008-05-13 12:42:17 -0500 (Tue, 13 May 2008) | 2 lines
Re-introduce proper error handling that was removed in recent commits.
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r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) | 11 lines
Merged revisions 115884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) | 3 lines
If the socket dies (read returns 0=EOF), return immediately.
(Closes issue #12637)
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r115939 | mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line
Add support for receiving calling party category
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r115941 | mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line
Need to clear calling_party_cat variable after we retrieve it
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r115945 | file | 2008-05-13 15:29:27 -0500 (Tue, 13 May 2008) | 12 lines
Merged revisions 115944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines
Use the right flag to open the audio in non-blocking.
(closes issue #12616)
Reported by: nicklewisdigiumuser
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r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) | 13 lines
Merged revisions 115990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines
Fix an issue that I noticed in autoservice while mmichelson and I were debugging
a different problem. I noticed that it was theoretically possible for two threads
to attempt to start the autoservice thread at the same time. This change makes the
process of starting the autoservice thread, thread-safe.
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r116039 | russell | 2008-05-13 16:18:55 -0500 (Tue, 13 May 2008) | 32 lines
Merged revisions 116038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines
Fix a deadlock involving channel autoservice and chan_local that was debugged
and fixed by mmichelson and me.
We observed a system that had a bunch of threads stuck in ast_autoservice_stop().
The reason these threads were waiting around is because this function waits to
ensure that the channel list in the autoservice thread gets rebuilt before the
stop() function returns. However, the autoservice thread was also locked, so
the autoservice channel list was never getting rebuilt.
The autoservice thread was stuck waiting for the channel lock on a local channel.
However, the local channel was locked by a thread that was stuck in the autoservice
stop function.
It turned out that the issue came down to the local_queue_frame() function in
chan_local. This function assumed that one of the channels passed in as an
argument was locked when called. However, that was not always the case. There
were multiple cases in which this channel was not locked when the function was
called. We fixed up chan_local to indicate to this function whether this channel
was locked or not. The previous assumption had caused local_queue_frame() to
improperly return with the channel locked, where it would then never get unlocked.
(closes issue #12584)
(related to issue #12603)
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r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May 2008) | 20 lines
Merged revisions 116088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines
A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS
is enabled in menuselect, the actual origin of channel locks is obscured
by the fact that all channel locks appear to happen in the function
ast_channel_lock(). This code change redefines ast_channel_lock to be a
macro which maps to __ast_channel_lock(), which then relays the proper
file name, line number, and function name information to the core lock
functions so that this information will be displayed in the case that
there is some sort of locking error or core show locks is issued.
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r116138 | mmichelson | 2008-05-13 19:20:05 -0500 (Tue, 13 May 2008) | 6 lines
Undo inadvertent changes to chan_skinny caused by the merging of urgent messaging
support.
Thanks to Damien Wedhorn for pointing out the problem.
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r116179 | oej | 2008-05-14 03:16:25 -0500 (Wed, 14 May 2008) | 2 lines
Doxygen formatting change only
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r116222 | oej | 2008-05-14 06:32:05 -0500 (Wed, 14 May 2008) | 2 lines
Adding comments
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r116223 | oej | 2008-05-14 06:37:21 -0500 (Wed, 14 May 2008) | 2 lines
Reformatting
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r116224 | oej | 2008-05-14 06:51:09 -0500 (Wed, 14 May 2008) | 2 lines
Formatting changes (coding guidelines) while thinking about something else...
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r116229 | oej | 2008-05-14 07:32:57 -0500 (Wed, 14 May 2008) | 5 lines
Add support for codec settings in originate via call file and manager.
This is to enable video and text in originated calls. Development sponsored
by Omnitor AB, Sweden. (http://www.omnitor.se)
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r116234 | oej | 2008-05-14 08:05:15 -0500 (Wed, 14 May 2008) | 11 lines
Merged revisions 116230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines
Accept text messages even with
Content-Type: text/plain;charset=Södermanländska
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r116237 | oej | 2008-05-14 08:37:07 -0500 (Wed, 14 May 2008) | 5 lines
Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
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r116239 | oej | 2008-05-14 09:03:42 -0500 (Wed, 14 May 2008) | 2 lines
Properly declare charset for text messages.
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r116240 | oej | 2008-05-14 09:16:51 -0500 (Wed, 14 May 2008) | 2 lines
Don't add linefeed on received MESSAGE
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r116297 | jpeeler | 2008-05-14 11:52:30 -0500 (Wed, 14 May 2008) | 3 lines
Fixed a few problems with multiparking: call not being parked in the correct parking spot, caller not being notified of parking spot position, and improperly hanging up the call during a transfer due to timing out (not providing the extension in which to transfer).
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r116298 | tilghman | 2008-05-14 11:53:23 -0500 (Wed, 14 May 2008) | 15 lines
Merged revisions 116296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) | 2 lines
Detect another way for a connection to have gone away.
(closes issue #12618)
Reported by: ctooley
Patches:
1.4-externalivr-test_fd.diff uploaded by ctooley (license 136)
trunk-externalivr-test_fd.diff uploaded by ctooley (license 136)
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r116350 | file | 2008-05-14 13:25:54 -0500 (Wed, 14 May 2008) | 4 lines
Make the ldap version setting work without having both version and protocol set.
(closes issue #12613)
Reported by: suretec
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r116353 | file | 2008-05-14 13:54:16 -0500 (Wed, 14 May 2008) | 12 lines
Merged revisions 116352 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines
Add linux-gnueabi in.
(closes issue #12529)
Reported by: tzafrir
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r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | 9 lines
Voicemail "* exit" should not require an exitcontext to be specified.
The behavior in 1.4 was that it would use the current context if an exitcontext existed.
(closes issue #12605)
Reported by: kenjreno
Patches:
12605-starexit.diff uploaded by qwell (license 4)
Tested by: file
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r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines
Merged revisions 116409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line
Document exitcontext in app_voicemail sample config
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r116461 | file | 2008-05-14 16:11:49 -0500 (Wed, 14 May 2008) | 6 lines
Add a missing context unlock.
(closes issue #12649)
Reported by: ys
Patches:
pbx.c.diff uploaded by ys (license 281)
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r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) | 15 lines
Merged revisions 116466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines
Avoid zombies when the channel exits before the AGI.
(closes issue #12648)
Reported by: gkloepfer
Patches:
20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
Tested by: gkloepfer
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r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines
Merged revisions 116463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines
Add ast_assert(), which can be used to handle fatal errors. It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)
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r116471 | file | 2008-05-14 16:54:03 -0500 (Wed, 14 May 2008) | 2 lines
Fix pedanticness.
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r116522 | mmichelson | 2008-05-14 17:15:12 -0500 (Wed, 14 May 2008) | 8 lines
Adding a new option to Chanspy(). The 'd' option allows for the spy to
press DTMF digits to switch between spying modes. Pressing 4 activates spy mode,
pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of
this feature overrides the normal operation of DTMF numbers.
This feature is courtesy of Switchvox.
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r116557 | rizzo | 2008-05-15 05:56:29 -0500 (Thu, 15 May 2008) | 10 lines
Use casts or intermediate variables to remove a number
of platform/compiler-dependent warnings when handing
struct timeval fields, both reading and printing them.
It is a lost battle to handle the different ways struct timeval
is handled on the various platforms and compilers, so try
to be pragmatic and go through int/long which are universally
supported.
................
r116590 | mmichelson | 2008-05-15 10:13:11 -0500 (Thu, 15 May 2008) | 8 lines
Prevent crashes from occurring due to a strcmp of a NULL pointer.
(closes issue #12661)
Reported by: jaroth
Patches:
urgentcompare.patch uploaded by jaroth (license 50)
................
r116592 | mmichelson | 2008-05-15 10:24:29 -0500 (Thu, 15 May 2008) | 9 lines
Modify externnotify to take the number of urgent voicemails as a final argument instead
of the string "Urgent"
(closes issue #12660)
Reported by: jaroth
Patches:
externnotify.patch uploaded by jaroth (license 50)
................
r116594 | mmichelson | 2008-05-15 10:40:29 -0500 (Thu, 15 May 2008) | 12 lines
When counting urgent messages when using IMAP storage, take into account that
the urgent messages are not in their own folder but are actually "flagged" messages
in the INBOX.
(closes issue #12659)
Reported by: jaroth
Patches:
urgentfolder_v2.patch uploaded by jaroth (license 50)
Tested by: jaroth
................
r116631 | tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines
Don't unload config on reload, when config has not changed.
(Closes issue #12652)
................
Added:
team/group/codec_bits/contrib/scripts/install_prereq
- copied unchanged from r116631, trunk/contrib/scripts/install_prereq
Removed:
team/group/codec_bits/contrib/scripts/postgres_cdr.sql
Modified:
team/group/codec_bits/ (props changed)
team/group/codec_bits/CHANGES
team/group/codec_bits/CREDITS
team/group/codec_bits/UPGRADE.txt
team/group/codec_bits/apps/app_chanspy.c
team/group/codec_bits/apps/app_externalivr.c
team/group/codec_bits/apps/app_jack.c
team/group/codec_bits/apps/app_queue.c
team/group/codec_bits/apps/app_skel.c
team/group/codec_bits/apps/app_voicemail.c
team/group/codec_bits/apps/app_waituntil.c
team/group/codec_bits/cdr/cdr_csv.c
team/group/codec_bits/cdr/cdr_pgsql.c
team/group/codec_bits/channels/chan_agent.c
team/group/codec_bits/channels/chan_alsa.c
team/group/codec_bits/channels/chan_h323.c
team/group/codec_bits/channels/chan_iax2.c
team/group/codec_bits/channels/chan_local.c
team/group/codec_bits/channels/chan_sip.c
team/group/codec_bits/channels/chan_unistim.c
team/group/codec_bits/channels/chan_zap.c
team/group/codec_bits/configs/queues.conf.sample
team/group/codec_bits/configs/voicemail.conf.sample
team/group/codec_bits/configure
team/group/codec_bits/configure.ac
team/group/codec_bits/contrib/init.d/rc.debian.asterisk
team/group/codec_bits/contrib/scripts/asterisk.ldap-schema
team/group/codec_bits/contrib/scripts/asterisk.ldif
team/group/codec_bits/doc/tex/channelvariables.tex
team/group/codec_bits/funcs/func_enum.c
team/group/codec_bits/funcs/func_speex.c
team/group/codec_bits/funcs/func_timeout.c
team/group/codec_bits/include/asterisk/app.h
team/group/codec_bits/include/asterisk/autoconfig.h.in
team/group/codec_bits/include/asterisk/channel.h
team/group/codec_bits/include/asterisk/dlinkedlists.h (props changed)
team/group/codec_bits/include/asterisk/enum.h
team/group/codec_bits/include/asterisk/frame.h
team/group/codec_bits/include/asterisk/lock.h
team/group/codec_bits/include/asterisk/rtp.h
team/group/codec_bits/include/asterisk/utils.h
team/group/codec_bits/main/Makefile
team/group/codec_bits/main/abstract_jb.c
team/group/codec_bits/main/app.c
team/group/codec_bits/main/asterisk.c
team/group/codec_bits/main/autoservice.c
team/group/codec_bits/main/channel.c
team/group/codec_bits/main/enum.c
team/group/codec_bits/main/features.c
team/group/codec_bits/main/frame.c
team/group/codec_bits/main/manager.c
team/group/codec_bits/main/pbx.c
team/group/codec_bits/main/rtp.c
team/group/codec_bits/main/sched.c
team/group/codec_bits/main/taskprocessor.c
team/group/codec_bits/main/udptl.c
team/group/codec_bits/main/utils.c
team/group/codec_bits/pbx/pbx_spool.c
team/group/codec_bits/res/res_agi.c
team/group/codec_bits/res/res_config_ldap.c
team/group/codec_bits/res/res_odbc.c
team/group/codec_bits/res/res_smdi.c
team/group/codec_bits/sample.call
team/group/codec_bits/sounds/Makefile
Propchange: team/group/codec_bits/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/codec_bits/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/group/codec_bits/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/codec_bits/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu May 15 14:48:22 2008
@@ -1,1 +1,1 @@
-/trunk:1-115481
+/trunk:1-116661
Modified: team/group/codec_bits/CHANGES
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/CHANGES?view=diff&rev=116662&r1=116661&r2=116662
==============================================================================
--- team/group/codec_bits/CHANGES (original)
+++ team/group/codec_bits/CHANGES Thu May 15 14:48:22 2008
@@ -42,6 +42,11 @@
quite helpful.
* Voicemail now permits a mailbox setting to wrap around from first to last
messages, if the "messagewrap" option is set to a true value.
+ * Voicemail now permits an external script to be run, for password validation.
+ The script should output "VALID" or "INVALID" on stdout, depending upon the
+ wish to validate or invalidate the password given. Arguments are:
+ "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
+ more details
* Dial has a new option: F(context^extension^pri), which permits a callee to
continue in the dialplan, at the specified label, if the caller hangs up.
* ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
@@ -54,6 +59,11 @@
to be spoken instead of the channel name or number. For more information on the
use of this option, issue the command "core show application ChanSpy" from the
Asterisk CLI.
+ * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
+ spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
+ words, if using the 'd' option, it is not possible to enter a number to append to
+ the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
+ change to whisper mode, and pressing 6 will change to barge mode.
SIP Changes
-----------
@@ -72,6 +82,8 @@
testing and problem reporting!
* Added ability to specify registration expiry time on a per registration basis in
the register line.
+ * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
+ lost packets.
IAX Changes
-----------
@@ -96,6 +108,12 @@
Dialplan function changes
-------------------------
* TIMEOUT() has been modified to be accurate down to the millisecond.
+ * ENUM*() functions now include the following new options:
+ - 'u' returns the full URI and does not strip off the URI-scheme.
+ - 's' triggers ISN specific rewriting
+ - 'i' looks for branches into an Infrastructure ENUM tree
+ - 'd' for a direct DNS lookup without any flipping of digits.
+ * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
@@ -155,6 +173,7 @@
* Originate now requires the Originate privilege and, if you want to call out
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.
+ * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
* New command: Atxfer. See doc/manager_1_1.txt for more details or
manager show command Atxfer from the CLI
@@ -406,6 +425,12 @@
voicemail boxes. The SMDI interface can also poll for MWI changes when some
outside entity is modifying the state of the mailbox (such as IMAP storage or
a web interface of some kind).
+ * Added the support for marking messages as "urgent." There are two methods to accomplish
+ this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
+ is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
+ the message as urgent after he has recorded a voicemail by following the voice instructions.
+ When listening to voicemails using VoiceMailMain urgent messages will be presented before other
+ messages
Queue changes
-------------
Modified: team/group/codec_bits/CREDITS
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/CREDITS?view=diff&rev=116662&r1=116661&r2=116662
==============================================================================
--- team/group/codec_bits/CREDITS (original)
+++ team/group/codec_bits/CREDITS Thu May 15 14:48:22 2008
@@ -19,6 +19,9 @@
John Todd, TalkPlus, Inc. and JR Richardson, Ntegrated Solutions. - for funding
the development of SIP Session Timers support.
+
+Omnitor AB, Gunnar Hellström, for funding work with videocaps, T.140 RED,
+originate with video/text and many more contributions.
=== WISHLIST CONTRIBUTERS ===
Jeremy McNamara - SpeeX support
Modified: team/group/codec_bits/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/UPGRADE.txt?view=diff&rev=116662&r1=116661&r2=116662
==============================================================================
--- team/group/codec_bits/UPGRADE.txt (original)
+++ team/group/codec_bits/UPGRADE.txt Thu May 15 14:48:22 2008
@@ -74,6 +74,8 @@
checking mailboxes for changes so that they can send MWI information to users.
Examples of situations that would require this option are web interfaces to
voicemail or an email client in the case of using IMAP storage.
+* The externnotify script should accept an additional (last) parameter
+ containing the number of urgent messages in the INBOX.
Applications:
Modified: team/group/codec_bits/apps/app_chanspy.c
URL: http://svn.digium.com/view/asterisk/team/group/codec_bits/apps/app_chanspy.c?view=diff&rev=116662&r1=116661&r2=116662
==============================================================================
--- team/group/codec_bits/apps/app_chanspy.c (original)
+++ team/group/codec_bits/apps/app_chanspy.c Thu May 15 14:48:22 2008
@@ -64,7 +64,8 @@
" - Dialing a series of digits followed by # builds a channel name to append\n"
" to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing\n"
" the digits '1234#' while spying will begin spying on the channel\n"
-" 'Agent/1234'.\n"
+" 'Agent/1234'. Note that this feature will be overriden if the 'd' option\n"
+" is used\n"
" Note: The X option supersedes the three features above in that if a valid\n"
" single digit extension exists in the correct context ChanSpy will\n"
" exit to it. This also disables choosing a channel based on 'chanprefix'\n"
@@ -73,6 +74,11 @@
" b - Only spy on channels involved in a bridged call.\n"
" B - Instead of whispering on a single channel barge in on both\n"
" channels involved in the call.\n"
+" d - Override the typical numeric DTMF functionality and instead\n"
+" use DTMF to switch between spy modes.\n"
+" 4 = spy mode\n"
+" 5 = whisper mode\n"
+" 6 = barge mode\n"
" g(grp) - Only spy on channels in which one or more of the groups \n"
" listed in 'grp' matches one or more groups from the\n"
" SPYGROUP variable set on the channel to be spied upon.\n"
@@ -126,6 +132,13 @@
" exit to it.\n"
" Options:\n"
" b - Only spy on channels involved in a bridged call.\n"
+" B - Instead of whispering on a single channel barge in on both\n"
+" channels involved in the call.\n"
+" d - Override the typical numeric DTMF functionality and instead\n"
+" use DTMF to switch between spy modes.\n"
+" 4 = spy mode\n"
+" 5 = whisper mode\n"
+" 6 = barge mode\n"
" g(grp) - Only spy on channels in which one or more of the groups \n"
" listed in 'grp' matches one or more groups from the\n"
" SPYGROUP variable set on the channel to be spied upon.\n"
@@ -162,19 +175,20 @@
;
enum {
- OPTION_QUIET = (1 << 0), /* Quiet, no announcement */
- OPTION_BRIDGED = (1 << 1), /* Only look at bridged calls */
- OPTION_VOLUME = (1 << 2), /* Specify initial volume */
- OPTION_GROUP = (1 << 3), /* Only look at channels in group */
- OPTION_RECORD = (1 << 4),
- OPTION_WHISPER = (1 << 5),
- OPTION_PRIVATE = (1 << 6), /* Private Whisper mode */
- OPTION_READONLY = (1 << 7), /* Don't mix the two channels */
- OPTION_EXIT = (1 << 8), /* Exit to a valid single digit extension */
- OPTION_ENFORCED = (1 << 9), /* Enforced mode */
- OPTION_NOTECH = (1 << 10), /* Skip technology name playback */
- OPTION_BARGE = (1 << 11), /* Barge mode (whisper to both channels) */
- OPTION_NAME = (1 << 12), /* Say the name of the person on whom we will spy */
+ OPTION_QUIET = (1 << 0), /* Quiet, no announcement */
+ OPTION_BRIDGED = (1 << 1), /* Only look at bridged calls */
+ OPTION_VOLUME = (1 << 2), /* Specify initial volume */
+ OPTION_GROUP = (1 << 3), /* Only look at channels in group */
+ OPTION_RECORD = (1 << 4),
+ OPTION_WHISPER = (1 << 5),
+ OPTION_PRIVATE = (1 << 6), /* Private Whisper mode */
+ OPTION_READONLY = (1 << 7), /* Don't mix the two channels */
+ OPTION_EXIT = (1 << 8), /* Exit to a valid single digit extension */
+ OPTION_ENFORCED = (1 << 9), /* Enforced mode */
+ OPTION_NOTECH = (1 << 10), /* Skip technology name playback */
+ OPTION_BARGE = (1 << 11), /* Barge mode (whisper to both channels) */
+ OPTION_NAME = (1 << 12), /* Say the name of the person on whom we will spy */
+ OPTION_DTMF_SWITCH_MODES = (1 << 13), /*Allow numeric DTMF to switch between chanspy modes */
} chanspy_opt_flags;
enum {
@@ -200,6 +214,7 @@
AST_APP_OPTION('X', OPTION_EXIT),
AST_APP_OPTION('s', OPTION_NOTECH),
AST_APP_OPTION_ARG('n', OPTION_NAME, OPT_ARG_NAME),
+ AST_APP_OPTION('d', OPTION_DTMF_SWITCH_MODES),
});
@@ -281,8 +296,22 @@
ast_mutex_t lock;
};
+static void change_spy_mode(const char digit, struct ast_flags *flags)
+{
+ if (digit == '4') {
+ ast_clear_flag(flags, OPTION_WHISPER);
+ ast_clear_flag(flags, OPTION_BARGE);
+ } else if (digit == '5') {
+ ast_clear_flag(flags, OPTION_BARGE);
+ ast_set_flag(flags, OPTION_WHISPER);
+ } else if (digit == '6') {
+ ast_clear_flag(flags, OPTION_WHISPER);
+ ast_set_flag(flags, OPTION_BARGE);
+ }
+}
+
[... 8600 lines stripped ...]
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