From a user point-of-view, this adds new CLI commands and Manager Actions to
better facilitate the reloading of queues and the resetting of their statistics.
The new CLI commands are the "queue reload" and "queue reset stats" commands.
The new manager actions are the QueueReload and QueueReset commands.
Review: http://reviewboard.digium.com/r/115
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines
Fix a potential crash situation when using IMAP voicemail
If calling into VoiceMailMain when using IMAP storage, it was
possible to crash Asterisk by hanging up the phone when prompted
for a voicemail mailbox. This patch fixes the issue.
While it may appear that this patch is superficial, it allows code
execution to continue to the failure case just below the IMAP_STORAGE
code block where this patch has been applied
(closes issue #14473)
Reported by: dwpaul
Patches:
voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a
"Thank you for your patience" sound when no position or holdtime
was actually announced.
This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.
There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.
(closes issue #14227)
Reported by: caspy
Patches:
14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.
The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely.
(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.
(closes issue #13673)
Reported by: howardwilkinson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines
Fix situations where queue members could be autopaused unexpectedly
Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.
(closes issue #14376)
Reported by: fiddur
Patches:
14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
(closes issue #14374)
Reported by: aragon
Patches:
14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
Reported by: oej
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.
To fix this problem, there are two measures taken
1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c
(closes issue #14260)
Reported by: ccesario
Patches:
14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines
Resolve a logic error that was causing Page() to crash when more than one
channel was specified.
(closes issue #14308)
Reported by: bluefox
Patches:
20090124__bug14308.diff.txt uploaded by seanbright (license 71)
Tested by: kc0bvu
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
(closes issue #14014)
Reported by: kebl0155
Patches:
14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
