Mark Michelson [Fri, 9 May 2008 21:22:42 +0000 (21:22 +0000)]
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
Brett Bryant [Fri, 9 May 2008 19:54:45 +0000 (19:54 +0000)]
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
Tilghman Lesher [Fri, 9 May 2008 17:28:06 +0000 (17:28 +0000)]
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)
Russell Bryant [Thu, 8 May 2008 19:17:04 +0000 (19:17 +0000)]
Merged revisions 115565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
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
........
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.
Russell Bryant [Wed, 7 May 2008 21:11:33 +0000 (21:11 +0000)]
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)
Russell Bryant [Wed, 7 May 2008 18:17:43 +0000 (18:17 +0000)]
Blocked revisions 115517 via svnmerge
........
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)
Separate verbose output from CLI output, by using a preamble.
(closes issue #12402)
Reported by: Corydon76
Patches:
20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14)
20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14)
Mark Michelson [Mon, 5 May 2008 22:14:06 +0000 (22:14 +0000)]
#execing the same file multiple times led to warning messages saying that the same file was
being #included twice. This was due to the fact that #exec created a temporary file which
was then #included. The name of the temporary file was the name of the #exec'd file, with
the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements
of the same file were reached in the same second, then the result was that the temporary files
would have duplicate names. To resolve this, the temporary file now has microsecond resolution
for the timestamp portion.
Don't consider a caller "handled" until the caller is bridged with
a queue member. There was too much of an opportunity for the member
to hang up (either during a delay, announcement, or overly long
agi) between the time that he answered the phone and the time when
he actually was bridged with the caller. The consequence of this
was that if the member hung up in that interval, then proper
abandonment details would not be noted in the queue log if the caller
were to hang up at any point after the member hangup.
Russell Bryant [Mon, 5 May 2008 19:50:24 +0000 (19:50 +0000)]
Merged revisions 115304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines
Avoid putting opaque="" in Digest authentication. This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)
Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile.
A taskprocessor is an object that has a name, a task queue, and an event processing thread. Modules reference a taskprocessor, push tasks into the taskprocessor as needed, and unreference the taskprocessor when the taskprocessor is no longer needed.
A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Clarify a comment that was, well, just wrong. It turns out that
ignoring the way that macros expand. Instead, I have clarified in the
comment why the macro will work even if the scheduler id for the
task to be deleted changes during the execution of the macro.
Tilghman Lesher [Thu, 1 May 2008 23:06:23 +0000 (23:06 +0000)]
Modify TIMEOUT() to be accurate down to the millisecond.
(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
Russell Bryant [Thu, 1 May 2008 19:05:36 +0000 (19:05 +0000)]
Merge changes from team/russell/smdi-msg-searching
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
'#' is another reserved character for URIs that also needs to be escaped.
(closes issue #10543)
Reported by: blitzrage
Patches:
20080418__bug10543.diff.txt uploaded by Corydon76 (license 14)
Russell Bryant [Thu, 1 May 2008 17:28:25 +0000 (17:28 +0000)]
Clarify the deprecation notice about Macro() to note that it will not be removed
for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.
Brett Bryant [Thu, 1 May 2008 16:57:19 +0000 (16:57 +0000)]
Add two new dialplan functions from libspeex for applying audio gain control
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
Mark Michelson [Wed, 30 Apr 2008 19:30:41 +0000 (19:30 +0000)]
Adding new configuration options to app_queue. This adds two new values
to announce-position, "limit" and "more," as well as a new option,
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.
Lock around variables retrieved, and copy the values, if they stay persistent,
since another thread could remove them.
(closes issue #12541)
Reported by: snuffy
Patches:
bug_12156_apps.diff uploaded by snuffy (license 35)
Several additional changes by me
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
Jeff Peeler [Wed, 30 Apr 2008 16:14:43 +0000 (16:14 +0000)]
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined
Jeff Peeler [Tue, 29 Apr 2008 22:54:14 +0000 (22:54 +0000)]
Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0.
Mark Michelson [Tue, 29 Apr 2008 21:07:36 +0000 (21:07 +0000)]
Patching app_chanspy to jibe better with what is documented. This allows for
a colon-delimited list of spygroups to be specified when calling the ChanSpy application
with the 'g' option. Prior to this, you could only specify a single group when using the
'g' option.
I also have upped the maximum number of spygroups to 128 and added a #define so that this
can be easily increased or decreased later.
Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.
(closes issue #12549)
Reported by: darren1713
Patches:
app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
(with modifications from me)
Tested by: putnopvut
Mark Michelson [Tue, 29 Apr 2008 17:56:13 +0000 (17:56 +0000)]
Since there is now a globally available function for saying someone's name, a LOT of
functions in app_directory can be removed since the ODBC-specific lookups are accomplished
within app_voicemail. This change greatly reduces the amount of lines in app_directory that
were solely for the purpose of looking up a name when ODBC_STORAGE is specified for voicemail.
This commit also makes the name-saying interruptable via DTMF.
Mark Michelson [Mon, 28 Apr 2008 22:38:07 +0000 (22:38 +0000)]
Adding a new option 'n' to app_chanspy. This option allows for the name of the spied-on
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
When we don't explicitly pass a path to the --with-tds configure option, we
may end up finding tds.h in /usr/local/include instead of /usr/include. If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.
Mark Michelson [Fri, 25 Apr 2008 22:24:32 +0000 (22:24 +0000)]
Adding a new option, 'B' to app_chanspy. This option allows the spy to
barge on the call. It is like the existing whisper option, except that
it allows the spy to talk to both sides of the conversation on which
he is spying.
This feature has existed in Switchvox, and this merges the functionality
into Asterisk.
projects / thirdparty / asterisk.git / log
