The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.
Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show".
Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.
Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].
Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.
(closes issue #17589)
Reported by: lmadsen
Patches:
issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
warning (at least with gcc 4.4.4):
netsock2.c:492: warning: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules
So we're back to using memcpy()...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to. This mirrors the expected behavior used in 1.4.
(closes issue #17444)
Reported by: corruptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.
(closes issue #16682)
Reported by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines
Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.
The issue here was that the frame created when adjusting for PLC had no offset
to its audio data. If this frame were translated to another format prior to
being sent out an RTP socket, all went well because the translation code would
put an appropriate offset into the frame. However, if the SLIN audio were not
translated before being sent out the RTP socket, bad things would happen.
Specifically, the ast_rtp_raw_write makes the assumption that the frame has
at least enough of an offset that it can accommodate an RTP header. This was
not the case. As such, data was being written prior to the allocation, likely
corrupting the data the memory allocator had written. Thus when the time came
to free the data, all hell broke loose. ....Well, Asterisk crashed at least.
The fix was just what one would expect. Offset the data in the frame by a reasonable
amount. The method I used is a bit odd since the data in the frame is 16 bit integers
and not bytes. I left a big ol' comment about it. This can be improved on if someone
is interested. I was more interested in getting the crash resolved.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Valgrind pointed out that attempting to get an IE value from an event that has
no IEs produces an invalid memory read past the end of the event. Thanks to
mmichelson for pointing the problem out to me and then testing the fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines
Fix Debian init script to not use -c.
When using the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so I'm
implementing the change. We now handle color displays properly.
(closes issue #16784)
Reported by: pabelanger
Patches:
20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
Tested by: pabelanger, tilghman
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Suppress the warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC
in file.c:waitstream_core().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
What I did not originally see in my previous commit was that even though the
next digit could be detected before the previous was considered ended, the
detection of the next digit effectively ends the detection of the previous.
Therefore, the length moves in lockstep with the digit, and no separate counter
is needed for the length alone.
(closes issue #17371)
Reported by: alecdavis
(closes issue #17474)
Reported by: kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ASTCFLAGS was specified with the make command, Makefile.rules was using
the specified value from the command line and not the one here, making it so this
flag would go missing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Actions (or failures to act) by external clients should not cause memory leaks
in Asterisk, especially when those continued leaks could cause Asterisk to
misbehave later.
(closes issue #17234)
Reported by: mav3rick
Patches:
20100510__issue17234.diff.txt uploaded by tilghman (license 14)
20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: mav3rick, davidw
(closes issue #17365)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines
Prevent CLI prompt from distorting output of lines shorter than the prompt.
Uses the VT100 method of clearing the line from the cursor position to the
end of the line: Esc-0K
(closes issue #17160)
Reported by: coolmig
Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines
Use sigaction for signals which should persist past the initial trigger, not signal.
If you call signal() in a Solaris signal handler, instead of just resetting
the signal handler, it causes the signal to refire, because the signal is not
marked as handled prior to the signal handler being called. This effectively
causes Solaris to immediately exceed the threadstack in recursive signal
handlers and crash.
(closes issue #17000)
Reported by: rmcgilvr
Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch breaks up every part of the sip registry string during
config parsing and removes all parsing from transmit_register().
Thanks to Nick_Lewis for contributing this patch!
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657)
chan_sip.c.patch uploaded by Nick Lewis (license 657)
chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657)
nicklewispatch.diff uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
Review: https://reviewboard.asterisk.org/r/628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The handling of the control subclass AST_CONTROL_READ_ACTION frame leaked
connected line string memory in __ast_read().
Also in __ast_read() the frame type switch should not have had a case for
AST_CONTROL_READ_ACTION. AST_CONTROL_READ_ACTION is not a frame type.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
__ast_channel_alloc_ap() has several points in the initialization of a new
channel structure where it could fail. Since the channel structure is now
an ao2 object, the destructor callback needs to be able to handle clean up
when the structure setup is incomplete.
Problems corrected:
1) Failing to setup the alertpipe would not unreference the structure but
free it directly. Doing this to an ao2_object is very bad.
2) File descriptors need to be initialized to -1 before a construction
failure could occur so the destructor will not close unopened descriptors.
3) The destructor needs to check that the string field has been
initialized before using any string field values. Crashes expected.
4) The destructor should not notify devstate if the device name is empty.
It is a waste of cycles and a couple ERROR log messages are generated.
Review: https://reviewboard.asterisk.org/r/675/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines
ast_callerid_parse() had a path that left name uninitialized.
Several callers of ast_callerid_parse() do not initialize the name
parameter before calling thus there is the potential to use an
uninitialized pointer.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Nick Lewis pointed out that the patch as committed wouldn't actually include
dynamic logger levels, which was missed by the other reviewers. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When decode_length returns the length there is a check to see if that
length is negative, if so the decode loop breaks as this means the
limit has been reached. The problem here is that length is an
unsigned int, so length can never be negative. This resulted in
an infinite loop.
(issue #17352)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #17235)
Reported by: frawd
Patches:
new_dtmf_dsp_len.patch uploaded by frawd (license 610)
20100518__issue17235.diff.txt uploaded by tilghman (license 14)
Tested by: frawd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now that Asterisk modules can dynamically create and destroy logger levels
on demand, it's useful to be able to configure a logger channel (console,
file, whatever) to be able to accept log messages from *all* levels, even
levels created dynamically. This patch adds support for this, by allowing
the '*' level name to be used in logger.conf.
Review: https://reviewboard.asterisk.org/r/663/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the keyword 'all' to the 'channel hangup request' CLI command
so that you can request all channels to be hungup without having to
restart Asterisk.
(closes issue #16009)
Reported by: moy
Patches:
hangup-all-rev-221688.patch uploaded by moy (license 222)
Tested by: moy, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines
Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences.
(closes issue #16749)
Reported by: dant
Patches:
dsp.c-bug16749-1.patch uploaded by dant (license 670)
Tested by: dant
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines
Fix logic error when checking for a devstate provider.
When using strsep, if one of the list of specified separators is not found,
it is the first parameter to strsep which is now NULL, not the pointer returned
by strsep.
This issue isn't especially severe in that the worst it is likely to do is waste
some cycles when a device with no '/' and no ':' is passed to ast_device_state.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sean Bright pointed out that we lost a set of newline characters in commit
190349 on a line I had recently changed. Yay for code review on commits.
(issue #17231, #10961)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines
Manager cookies are not compatible with RFC2109.
The Version field in the cookies we're setting contain quotes around the version
number which is not compatible with RFC2109 and breaks some implementations.
(closes issue #17231)
Reported by: ecarruda
Patches:
manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559)
manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559)
Tested by: ecarruda, russell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As pointed out by 'akshayb' on #asterisk-dev, the code here called a list of
bucket entries a 'bucket', and the entries within the bucket were called
'bucket_list'. This made the code very hard to understand without reading
all of it... so I've renamed 'bucket_list' to 'bucket_entry' to clarify the
purpose of the structure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop. Additionally, this adds a res_timing interface, using kqueue timers.
Review: https://reviewboard.asterisk.org/r/543/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This bug surfaced in 1.6.2 and does not affect code in any other released
version of Asterisk. It manifested itself as SIP qualify not happening when
it should, causing peers to go unreachable. This was debugged down to scheduler
entries sometimes not getting executed when they were supposed to, which was in
turn caused by an error in the heap code.
The problem only sometimes occurs, and it is due to the logic for removing an entry
in the heap from an arbitrary location (not just popping off the top). The scheduler
performs this operation frequently when entries are removed before they run (when
ast_sched_del() is used).
In a normal pop off of the top of the heap, a node is taken off the bottom,
placed at the top, and then bubbled down until the max heap property is restored
(see max_heapify()). This same logic was used for removing an arbitrary node
from the middle of the heap. Unfortunately, that logic is full of fail. This
patch fixes that by fully restoring the max heap property when a node is thrown
into the middle of the heap. Instead of just pushing it down as appropriate, it
first pushes it up as high as it will go, and _then_ pushes it down.
Lastly, fix a minor problem in ast_heap_verify(), which is only used for
debugging. If a parent and child node have the same value, that is not an
error. The only error is if a parent's value is less than its children.
A huge thanks goes out to cappucinoking for debugging this down to the scheduler,
and then producing an ast_heap test case that demonstrated the breakage. That
made it very easy for me to focus on the heap logic and produce a fix. Open source
projects are awesome.
(closes issue #16936)
Reported by: ib2
Tested by: cappucinoking, crjw
(closes issue #17277)
Reported by: cappucinoking
Patches:
heap-fix.rev2.diff uploaded by russell (license 2)
Tested by: cappucinoking, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines
Protect against overflow, when calculating how long to wait for a frame.
(closes issue #17128)
Reported by: under
Patches:
d.diff uploaded by under (license 914)
........
r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines
Add a tiny corner case to the previous commit
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have some functions inside the AstData API to get the tree
in XML form, but it is not required at the moment to compile
asterisk and we can disable that part of the API if we don't have
libxml2 support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Matthew Nicholson
Tilghman Lesher
Mark Michelson
Richard Mudgett
Eliel C. Sardanons
Paul Belanger
Tzafrir Cohen
David Vossel
Jeff Peeler
Terry Wilson
Kevin P. Fleming
Leif Madsen