Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines
Solaris compatibility fixes
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288637 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286115 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285961 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285930 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279946 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
There was a rather large syntax error that should have caused ALL versions of GNU make to fail.
I don't know how it worked.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279657 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines
Merged revisions 278167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@278273 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines
Merged revisions 277738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines
Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not.
(closes issue #17616)
Reported by: pprindeville
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@277776 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines
Merged revisions 275021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275023 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273831 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Also, update existing test modules that were already in this branch but had
been converted to the unit test API in trunk.
Review: https://reviewboard.asterisk.org/r/748/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272531 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326)
Reported by: kenner
Tested by: mnicholson, kenner
Review: https://reviewboard.asterisk.org/r/693/
........
This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271691 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010) | 18 lines
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines
Fix crash when parsing some heavily nested statements in AEL on reload.
Due to the recursion used when compiling AEL in gen_prios, all the stack space
was being consumed when parsing some AEL that contained nesting 13 levels deep.
Changing a few large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used.
(closes issue #16053)
Reported by: diLLec
Tested by: jpeeler
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271484 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r269417 | russell | 2010-06-09 16:11:43 -0500 (Wed, 09 Jun 2010) | 6 lines
Resolve an invalid memory read on an event.
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@269418 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This is what caused a bunch of tests to fail on 1.6.2. They expected a console
channel driver, but chan_oss was failing to load.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@268815 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r267862 | tilghman | 2010-06-03 21:58:55 -0500 (Thu, 03 Jun 2010) | 5 lines
As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
........
r267877 | tilghman | 2010-06-03 22:20:47 -0500 (Thu, 03 Jun 2010) | 8 lines
As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory.
(closes issue #16912)
Reported by: michaelevdokimov
Patches:
asterisk.patch uploaded by michaelevdokimov (license 997)
Tested by: michaelevdokimov
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267883 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r267775 | tilghman | 2010-06-03 20:20:17 -0500 (Thu, 03 Jun 2010) | 14 lines
Merged revisions 267759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) | 7 lines
Make the default install path appear to be /usr on Linux, instead of /usr/local.
Also, reorganize the options, so that they're more alphabetical.
(closes issue #17013)
Reported by: klaus3000
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267787 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267527 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r267492 | mmichelson | 2010-06-03 12:09:11 -0500 (Thu, 03 Jun 2010) | 6 lines
Remove unnecessary code relating to PLC.
The logic for handling generic PLC is now handled in ast_write in
channel.c instead of in translation code.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267507 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r267065 | jpeeler | 2010-06-02 12:29:35 -0500 (Wed, 02 Jun 2010) | 12 lines
Fix infinite loop when loading codec speex
This changes the sample slinear frame data to contain non-zero data so that
translation calculations for speex works when preprocessing and VAD is turned
on. The encoder expects samples to be returned, but when attempted with the
mentioned two options and silent sample frames everything was discarded.
(closes issue #17240)
Reported by: seandarcy
Review: https://reviewboard.asterisk.org/r/682/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@267073 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r265747 | tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
Use configure to determine the prefixes and include directories properly.
This ensures cross-platform compatibility, even among Linux distributions,
which don't always put headers in the same place.
(closes issue #17391)
Reported by: loloski
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265748 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
Review: https://reviewboard.asterisk.org/r/639/
........
r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
Merge the rest of the FullyBooted patch
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265521 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May 2010) | 15 lines
Merged revisions 265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines
Don't hang up on a queue caller if the file we attempt to play does not exist.
This also fixes a documentation mistake in file.h that made my original attempt
to correct this problem not work correctly.
(closes issue #17061)
Reported by: RoadKill
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265091 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r265000 | mmichelson | 2010-05-21 11:54:21 -0500 (Fri, 21 May 2010) | 9 lines
Merged revisions 264999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines
Fix grammatical error in comment.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265001 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r264997 | mmichelson | 2010-05-21 11:44:27 -0500 (Fri, 21 May 2010) | 38 lines
Merged revisions 264996 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@264998 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines
Fix transcode_via_sln option with SIP calls and improve PLC usage.
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@264453 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) | 24 lines
Merged revisions 264248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines
Internal timing is now on by default, if you're using DAHDI 2.3 or above.
The reason for ensuring DAHDI 2.3 or above is that this version ensures that
a timer is always available, whereas in previous versions, it was possible
for DAHDI to be loaded, but have no drivers to actually generate timing. If
internal_timing was turned on in this circumstance, a complete lack of audio
would result. This is the reason why internal_timing was not on by default.
However, now that DAHDI ensures the availability of a timer, there is no
reason for this setting to be off (and in fact, it solves a great many initial
user problems).
(closes issue #15932)
Reported by: dimas
Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@264250 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 May 2010) | 7 lines
Move cause 200 to cause 26, as specified in Q.850.
Also cleanup the formatting and add a few more that seem like good candidates.
(closes issue #16157)
Reported by: wimpy
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@262516 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 May 2010) | 5 lines
Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS.
(closes issue #17309)
Reported by: stuarth
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@262105 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r261913 | tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 lines
Use the detected pthread building flags in every place, instead of hardcoding -lpthread.
We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads. This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.
(closes issue #17303)
Reported by: stuarth
Patches:
20100507__issue17303.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@261916 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@260884 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines
Merged revisions 260049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines
Fixes crash in audiohook_write_list
The middle_frame in the audiohook_write_list function was
being freed if a audiohook manipulator returned a failure.
This is incorrect logic. This patch resolves this and
adds detailed descriptions of how this function should work
and why manipulator failures must be ignored.
(closes issue #17052)
Reported by: dvossel
Tested by: dvossel
(closes issue #16196)
Reported by: atis
Review: https://reviewboard.asterisk.org/r/623/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@260051 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r257560 | tilghman | 2010-04-15 16:26:19 -0500 (Thu, 15 Apr 2010) | 13 lines
Merged revisions 257544 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines
Allow application options with arguments to contain parentheses, through a variety of escaping techniques.
Fixes SWP-1194 (ABE-2143).
Review: https://reviewboard.asterisk.org/r/604/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@257597 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) | 2 lines
Mac OS X does not support comparing a mutex to its initializer. Create a test for this.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@256373 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) | 7 lines
Fix DEBUG_THREADS build on Darwin.
(closes issue #16828)
Reported by: oej
Patches:
20100331__issue16828.diff.txt uploaded by tilghman (license 14)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@255816 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Take 2, without ABI breakage this time.
Review: https://reviewboard.asterisk.org/r/588/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@254770 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r254553 | mmichelson | 2010-03-25 12:42:36 -0500 (Thu, 25 Mar 2010) | 11 lines
Merged revisions 254552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines
Add doxygen for acl.h
Review: https://reviewboard.asterisk.org/r/528
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@254556 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This re-renames ast_rtp_update_source to ast_rtp_new_source
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@253158 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@252137 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration
When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249895 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
Properly document voicemail API documents. Also fix a crash reported via the -dev list.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249407 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines
Fix two problems in ast_str functions found while writing a unit test.
1. The documentation for ast_str_set and ast_str_append state that
the max_len parameter may be -1 in order to limit the size of the
ast_str to its current allocated size. The problem was that the max_len
parameter in all cases was a size_t, which is unsigned. Thus a -1 was
interpreted as UINT_MAX instead of -1. Changing the max_len parameter
to be ssize_t fixed this issue.
2. Once issue 1 was fixed, there was an off-by-one error in the case
where we attempted to write a string larger than the current allotted
size to a string when -1 was passed as the max_len parameter. When trying
to write more than the allotted size, the ast_str's __AST_STR_USED was
set to 1 higher than it should have been. Thanks to Tilghman for quickly
spotting the offending line of code.
Oh, and the unit test that I referenced in the top line of this commit
will be added to reviewboard shortly. Sit tight...
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247337 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines
Add some clarifying documentation to the ast_str_set and ast_str_append functions.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@246989 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r246030 | tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 lines
Solaris doesn't like outputting a NULL to a %s in format strings.
Detect all platforms that don't like that, either, and ensure that when documentation is
missing, we pass a non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689)
Reported by: bklang
Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/497/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@246199 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field
AST-2010-001
(closes issue #16634)
Reported by: krn
(closes issue #16724)
Reported by: barthpbx
(closes issue #16517)
Reported by: bklang
(closes issue #16485)
Reported by: elsto
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@244445 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r242521 | tilghman | 2010-01-24 00:40:31 -0600 (Sun, 24 Jan 2010) | 15 lines
Merged revisions 242520 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@242522 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r238635 | dvossel | 2010-01-08 13:39:30 -0600 (Fri, 08 Jan 2010) | 22 lines
fixes AUDIOHOOK_INHERIT regression
During the process of removing an audiohook from one channel
and attaching it to another the audiohook's status is updated
to DONE and then back to whatever it was previously. Typically
updating the status after setting it to DONE is not a good idea
because DONE can trigger unrecoverable audiohook destruction
events... because of this a conditional check was added to
audiohook_update_status to explicitly prevent the audiohook
from ever changing after being set to DONE. It was this check
that prevented audiohook inherit from work properly though.
Now ast_audiohook_move_by_source is treated as a special exception,
as the audiohook must be returned to its previous status after
attaching it to the new channel. This is only a safe operation
because the audiohook's lock is held the entire time, otherwise
this could cause trouble.
(closes issue #16522)
Reported by: corruptor
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@238637 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines
Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files.
(closes issue #16251)
Reported by: asgaroth
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@238499 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines
Merged revisions 237405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@237409 f38db490-d61c-443f-a65b-d21fe96a405b
|