removed. No telling when it happened. Anyway, it's back in now
and works properly.
(Based on issue reported on mailing list)
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them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines
davidw pointed out that the holdtime calculation used by
app_queue does not use "boxcar" filtering as the comments
say. The term "boxcar" means that the number of samples used
to calculate stays constant, with new samples replacing the
oldest ones. The queue holdtime calculation uses all holdtime
samples collected since the queue was loaded, so the comment
has been changed to be accurate.
(closes issue #12781)
Reported by: davidw
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ys
Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.
I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c
I did a simple sanity test to make sure the code doesn't
mess things up in general.
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long-term use. Instead use the heap. I can't believe this
never happened *once* in my developer branch when I was testing.
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net result of this work is that attended transfers made
by queue members will now show up in the queue_log as a
TRANSFER message instead of COMPLETECALLER as it had been.
As far as the details go, I created a datastore which is
attached to the calling channel just prior to when the caller
is bridged with the queue member. If the calling channel
is masqueraded, then during the "fixup" portion, the TRANSFER
will be logged and the datastore will be removed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
........
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
........
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This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
where the entered phone number is checked.
You can now define a set of extensions/exten patterns that describe
valid phone numbers. PrivacyManager will check that context for a match
with the given phone number.
This way you get better control. For example people blindly hitting
10 digits just to get past privacymanager
Example line in extensions.conf:
exten => incoming,n,PrivacyManager(3,10,,route-outgoing)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
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a dynamic realtime queue member is added to the queue, and the
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.
(closes issue #12774)
Reported by: atis
Patches:
queue_log_rt_members.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being
distributed under the LGPL, so we can move this module into the main tree.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines
When joinempty=strict, it only failed on join if there were busy members. If
all members were logged out OR paused, then it (incorrectly) let callers join
the queue.
(closes issue #12451)
Reported by: davidw
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines
(closes issue #10668)
(closes issue #11721)
(closes issue #12726)
Reported by: arkadia
Tested by: murf
These changes:
1. revert the changes made via bug 10668;
I should have known that such changes,
even tho they made sense at the time,
seemed like an omission, etc, were actually
integral to the CDR system via forkCDR.
It makes sense to me now that forkCDR didn't
natively end any CDR's, but rather depended
on natively closing them all at hangup time
via traversing and closing them all, whether
locked or not. I still don't completely
understand the benefits of setvar and answer
operating on locked cdrs, but I've seen
enough to revert those changes also, and
stop messing up users who depended on that
behavior. bug 12726 found reverting the changes
fixed his changes, and after a long review
and working on forkCDR, I can see why.
2. Apply the suggested enhancements proposed
in 10668, but in a completely compatible
way. ForkCDR will behave exactly as before,
but now has new options that will allow some
actions to be taken that will slightly
modify the outcome and side-effects of
forkCDR. Based on conversations I've had
with various people, these small tweaks
will allow some users to get the behavior
they need. For instance, users executing
forkCDR in an AGI script will find the
answer time set, and DISPOSITION set,
a situation not covered when the routines
were first written.
3. A small problem in the cdr serializer
would output answer and end times even
when they were not set. This is now
fixed.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
retrieving the value of a channel variable. This covers app_queue.
This commit also incorporates a logical change. Previously, if MixMonitor
is to be used to record the call, all the arguments were parsed first. Then
the MixMonitor app would be located. Now the order of these operations has
been swapped. Now the app is located first so that we only go through the
work of parsing the arguments if the app was found.
(closes issue #12742)
Reported by: snuffy
Patches:
bug_12742.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines
Russell noted to me that in the case that separate threads use their
own addressing system, the fix I made for issue 12376 does not guarantee
uniqueness to the datastores' uids. Though I know of no system that works
this way, I am going to change this right now to prevent trying to track
down some future bug that may occur and cause untold hours of debugging
time to track down.
The change involves using a global counter which increases with each new
chanspy_ds which is created. This guarantees uniqueness.
........
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines
Add a unique id to the datastore allocated in app_chanspy since
it is possible that multiple spies may be listening to the same
channel.
(closes issue #12376)
Reported by: DougUDI
Patches:
12376_chanspy_uid.diff uploaded by putnopvut (license 60)
Tested by: destiny6628
(closes issue #12243)
Reported by: atis
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
likely that after the event is freed, we no longer refer to valid memory.
(closes issue #12712)
Reported by: tomo1657
Patches:
12712.patch uploaded by putnopvut (license 60)
Tested by: tomo1657
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you give chanisavail a list of channels, it will only
return the first available channel.
When this option is set, it will return all the available
channels from the given list.
(closes issue #12248)
Reported by: dagmoller
Patches:
app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license 436)
- major changes by me because russellb pointed out some buffer overflows
and codeguideline issues.
Converted it all to the ast_str_* api
Tested by: dagmoller, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #12705)
Reported by: ctooley
Patches:
new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
new_externalivr_documentation.diff uploaded by ctooley (license 136)
and a few additional fixes by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to query the database for the member and instead using a cached
uniqueid.
Special thanks to atis for creating this and for keeping it up
to date with necessary changes
(closes issue #11896)
Reported by: atis
Patches:
realtime_uniqueid_v6.patch uploaded by atis (license 242)
Tested by: atis
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May 2008) | 13 lines
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.
(closes issue #12561)
Reported by: ablackthorn
........
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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.
(closes issue #10991)
Reported by: slavon
Patches:
app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue #12497)
Reported by: jsmith
Patches:
app_chanspy_multiple_groups_v2.patch uploaded by jsmith (license 15)
Tested by: atis, jvandal
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines
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
........
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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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.
(AST-32)
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Initially, this was to be a new feature, with a patch from Switchvox,
but after discussions, it was noted that this feature already existed in trunk.
The resulting discussions ended in a comment that was along the lines of
"the patch provided here is a lot smaller than what is already in trunk,
because it doesn't create a new application and duplicate existing code"
It was decided that these two applications could be easily merged to reduce
code duplication. SO, that's what this does.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
if using IMAP storage for voicemail. The comment will be recorded and attached
as a second attachment in addition to the original message. This will be invoked
if you choose to prepend a message the way you would with file or ODBC storage
(closes issue #12028)
Reported by: jaroth
Patches:
forward_with_comment_v2.patch uploaded by jaroth (license 50)
Tested by: jaroth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) | 10 lines
Fix an issue that caused getting the correct next channel to not always work.
Also, remove setting the amount of time to wait for a digit from 5 seconds back
down to 1/10 of a second. I believe this was so the beep didn't get played over
and over really fast, but a while back I put in another fix for that issue.
(closes issue #12498)
Reported by: jsmith
Patches:
app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15)
........
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This fixes a bug that was thought to be fixed already.
app_voicemail, if using IMAP_STORAGE, has a problem because
the IMAP header files include syslog.h, which define LOG_WARNING
and LOG_DEBUG to be different than what Asterisk uses for those
same macros. This was "fixed" in the past by including all the
IMAP header files prior to including asterisk.h. This fix worked...
unless you were to try to compile with MALLOC_DEBUG. MALLOC_DEBUG
prepends the inclusion of astmm.h to every file, which means that no
matter what order the includes are in in app_voicemail, the unexpected
values for LOG_WARNING and LOG_DEBUG will be in place.
The action taken for this fix was to define AST_LOG_* macros in addition
to the LOG_* macros already defined. These new macros are used in app_voicemail.c,
logger.h, and astobj.h right now, and their use will be encouraged in the future.
In consideration of those who have written third-party modules which use
the LOG_* macros, these will NOT be removed from the source, however future use
of these macros is discouraged.
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This makes IMAP_STORAGE include the proper headers if you
have specified the "system" option for --with-imap when running
the configure script and your IMAP-related headers exist in
/usr/include/c-client.
This change is due to a hasty merge of a 1.4 change I made.
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a custom client name. Using the channel name is still the default. This was done
at the request of Jared Smith.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr 2008) | 9 lines
Declaration of the peer channel in this scope was making it so the peer variable defined
in the outer scope was never set properly, therefore making iterating through the channel
list always restart from the beginning. This bug would have affected anyone who called
chanspy without specifying a first argument.
(closes issue #12461)
Reported by: stever28
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r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | 8 lines
Allow autofill to work in the general section of queues.conf.
Additionally, don't try to (re)set options when they have empty values in realtime (all unset columns would have an empty value).
(closes issue #12445)
Reported by: atis
Patches:
12445-autofill.diff uploaded by qwell (license 4)
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r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines
If the datastore has been moved to another channel due to a masquerade, then
freeing the datastore here causes an eventual double free when the new channel
hangs up. We should only free the datastore if we were able to successfully remove
it from the channel we are referencing (i.e. the datastore was not moved).
(closes issue #12359)
Reported by: pguido
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The variable name "flag" to distinguish between whether a message is being forwarded or
is new is not a helpful name. The newly added doxygen documentation to app_voicemail is
tremendously helpful, but I still just...hate this variable name. I think is_new_message
is more indicative of what its purpose is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(same patch as before, I just split this part out)
(close issue #12326)
Reported by: travishein
Patches:
app_voicemail_code_documentation.patch uploaded by travishein (license 385)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem was that when the refcount on the queue hit 0, the destructor was
called, and inside the destructor, another function was called which would increase
the refcount back to 1 again and then decrease it again back to 0 for every member
in the queue. This meant that the destructor was being recursively called, leading
to a double free of the queue. This is now fixed by making sure to unlink the
queue from the queues container prior to the final unref of the queue.
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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines
These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.
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r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines
Add a lock to the vm_state structure and use the lock around mail_open calls
to prevent concurrent access of the same mailstream. This, along with trunk's
ability to configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage.
(closes issue #10487)
Reported by: ewilhelmsen
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1) Number of digits to enter can now be configured
2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue #7151)
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r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines
This patch makes it so that all queue member status changes are handled through device state
code. This removes several problems people were seeing where their queue members would get into
an "unknown" state. Huge props go to atis on this one since he was the one who found the code
section that was causing the problem and proposed the solution. I just wrote what he suggested :)
(closes issue #12127)
Reported by: atis
Patches:
12127v3.patch uploaded by putnopvut (license 60)
Tested by: atis, jvandal
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.
(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines
Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup(). The concept is exactly like the
fixup callback that is used in the channel technology interface. This callback
gets called when the owning channel changes due to a masquerade. Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.
(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)
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r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines
Fix a couple uses of sprintf. The second one could actually cause an overflow
of a stack buffer. It's not a security issue though, it only depends on your
configuration.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ctooley
Patches:
eivr_tcp_generic.patch uploaded by jpeeler (license 325)
This change adds the ability to communicate over a TCP socket instead of forking a child process.
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r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines
(closes issue #12187, reported by atis, fixed by me after some brainstorming
on the issue with mmichelson)
- Update copyright info on app_chanspy.
- Fix a race condition that caused app_chanspy to crash. The issue was that
the chanspy datastore magic that was used to ensure that spyee channels did
not disappear out from under the code did not completely solve the problem.
It was actually possible for chanspy to acquire a channel reference out of
its datastore to a channel that was in the middle of being destroyed. That
was because datastore destruction in ast_channel_free() was done near the
end. So, this left the code in app_chanspy accessing a channel that was
partially, or completely invalid because it was in the process of being free'd
by another thread. The following sort of shows the code path where the race
occurred:
=============================================================================
Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
--------------------------------------||-------------------------------------
ast_channel_free() ||
- remove channel from channel list ||
- lock/unlock the channel to ensure ||
that no references retrieved from ||
the channel list exist. ||
--------------------------------------||-------------------------------------
|| channel_spy()
- destroy some channel data || - Lock chanspy datastore
|| - Retrieve reference to channel
|| - lock channel
|| - Unlock chanspy datastore
--------------------------------------||-------------------------------------
- destroy channel datastores ||
- call chanspy datastore d'tor ||
which NULL's out the ds' || - Operate on the channel ...
reference to the channel ||
||
- free the channel ||
||
|| - unlock the channel
--------------------------------------||-------------------------------------
=============================================================================
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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
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r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines
Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue #12136)
Reported by: aragon
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those XXX comments from the code.
The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are
not set, so there's no need to explicitly check them again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was the way that locks are referenced, since the old 1.2 names were still used
in the comments.
(closes issue #11997)
Reported by: snuffy
Patches:
bug_11997_queue_doxy.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
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r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb 2008) | 6 lines
When using autofill, members who are in use should be counted towards the
number of available members to call if ringinuse is set to yes.
Thanks to jmls who brought this issue up on IRC
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008) | 10 lines
This patch fixes some pretty significant problems with how app_chanspy handles
pointers to channels that are being spied upon. It was very likely that a
crash would occur if the channel being spied upon hung up. This was because
the current ast_channel handling _requires_ that the object is locked or else
it could disappear at any time (except in the owning channel thread). So, this
patch uses some channel datastore magic on the spied upon channel to be able to
detect if and when the channel goes away.
(closes issue #11877)
(patch written by me, but thanks to kpfleming for the idea, and to file for review)
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r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb 2008) | 8 lines
Clear up confusion when viewing the QUEUE_WAITING_COUNT of a
"dead" realtime queue. Since from the user's perspective, the queue
does exist, we shouldn't tell them we couldn't find the queue. Instead
since it is a dead queue, report a 0 waiting count
This issue was brought up on IRC by jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #11553)
Reported by: johan
Patches:
UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
CHANGES.channelredirect.patch uploaded by johan (license 334)
app_channelredirect-20080219.patch uploaded by johan (license 334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New feature: Add the 'e' option, which takes as an argument a list of
interfaces separated by colons. This way, you will only be able to spy
on this limited list of interfaces.
Bug fix: change some pointer checks to ast_strlen_zero so that spying
would work properly even if no channel was specified as the first argument
to chanspy.
(closes issue #10072)
Reported by: xmarksthespot
Patches:
bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by xmarksthespot (license 16)
Tested by: xmarksthespot, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
not be loaded from realtime queues. This commit fixes that.
Thanks to jmls for pointing this problem out to me on IRC.
This also contains some changes to S_OR where it should be used. Thanks to Qwell for pointing
these out.
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This is done in a backward compat way.
If the "default" key for ffwd/rew is used for another option (such as stop), the "default" is removed.
(closes issue #11754)
Reported by: johan
Patches:
app_controlplayback.c.option3.patch uploaded by johan (license 334)
Tested by: johan, qwell
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r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan 2008) | 5 lines
Fix a logic error with regards to autofill. Prior to this change, it was possible
for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting
to call a member. This change fixes this.
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r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | 8 lines
ChanSpy issues a beep when it starts at the beginning of a list of channels to
potentially spy on. However, if there were no matching channels, it would beep
at you over and over, which is pretty annoying. Now, it will only beep once in
the case that there are no channels to spy on, but it will still beep again once
it reaches the beginning of the channel list again.
(closes issue #11738, patched by me)
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r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) | 8 lines
When we reset the password via an external command, we should also reset the
password stored in the in-memory list, too (otherwise it doesn't really take
effect).
(closes issue #11809)
Reported by: davetroy
Patches:
fix_externpass.diff uploaded by davetroy (license 384)
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so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:
"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.
1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."
This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.
Huge thanks to dimas for this wonderful submission.
(closes issue #11744)
Reported by: dimas
Patches:
dir3.patch uploaded by dimas (license 88)
Tested by: putnopvut, dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines
Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.
Thanks to John Bigelow for pointing this error out to me.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
option tells JACK not to start jackd automatically if it is not already
running. Otherwise, the default is that jackd will get started for you if
it isn't running already.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines
Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark Michelson
Tilghman Lesher
Brett Bryant
Steve Murphy
Chris Tooley
Sean Bright
Jeff Peeler
Russell Bryant
Michiel van Baak
Jason Parker
Jim Dixon
Luigi Rizzo
Claude Patry
Joshua Colp
Dwayne M. Hubbard
Terry Wilson
Kevin P. Fleming
Olle Johansson