[asterisk-commits] eliel: branch eliel/data_api_providers_gsoc2010 r269932 - in /team/eliel/data...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jun 11 09:24:18 CDT 2010
Author: eliel
Date: Fri Jun 11 09:24:07 2010
New Revision: 269932
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=269932
Log:
Merged revisions 267041,267065,267093,267096-267097,267138,267181,267261,267303,267305,267350,267352,267399,267445,267490,267492,267537,267622,267624,267669,267714,267775,267819,267862-267863,267877,267928,267972,268051,268127,268205,268281,268321,268395,268417,268454,268456,268495,268534,268578,268653,268690,268731,268734,268773-268774,268817-268818,268857,268894,268896,268933,268969,268988,269007-269008,269083,269119,269153,269187,269196-269205,269238,269271,269307-269308,269346,269417,269486,269497,269569,269602,269636,269707,269711,269749,269822,269889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r267041 | pabelanger | 2010-06-02 13:25:05 -0400 (Wed, 02 Jun 2010) | 14 lines
Merged revisions 267009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun 2010) | 7 lines
Cleanup error/warning messages in AEL2 parser
(closes issue #16684)
Reported by: Silmaril
Patches:
patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
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r267065 | jpeeler | 2010-06-02 13:29:35 -0400 (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/
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r267093 | russell | 2010-06-02 13:57:39 -0400 (Wed, 02 Jun 2010) | 2 lines
Silence a compiler warning.
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r267096 | rmudgett | 2010-06-02 14:10:15 -0400 (Wed, 02 Jun 2010) | 38 lines
Generic Advice of Charge.
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/
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r267097 | mmichelson | 2010-06-02 14:13:18 -0400 (Wed, 02 Jun 2010) | 9 lines
Prevent use of uninitialized values.
Two struct sockaddr_ins are created when applying directmedia
host access rules. The addresses of these are passed to the RTP
engine to be filled in. However, the RTP engine inspects the fields
of the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
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r267138 | russell | 2010-06-02 14:53:38 -0400 (Wed, 02 Jun 2010) | 4 lines
Add a CLI command that blocks until Asterisk has fully booted.
Review: https://reviewboard.asterisk.org/r/684/
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r267181 | dvossel | 2010-06-02 15:33:56 -0400 (Wed, 02 Jun 2010) | 2 lines
Update CHANGES and aoc help doc to reflect AOC additions
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r267261 | rmudgett | 2010-06-02 17:05:32 -0400 (Wed, 02 Jun 2010) | 20 lines
Add ETSI Call Waiting support.
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
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r267303 | russell | 2010-06-02 17:41:54 -0400 (Wed, 02 Jun 2010) | 6 lines
Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been specified.
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.
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r267305 | russell | 2010-06-02 17:44:06 -0400 (Wed, 02 Jun 2010) | 2 lines
Fix a build error on mac.
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r267350 | rmudgett | 2010-06-02 18:28:58 -0400 (Wed, 02 Jun 2010) | 9 lines
Add ETSI Malicious Call ID support.
Add the ability to report malicious callers as an AMI event in the call
event class.
Relevant specification: EN 300 180
Review: https://reviewboard.asterisk.org/r/576/
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r267352 | russell | 2010-06-02 18:46:37 -0400 (Wed, 02 Jun 2010) | 7 lines
try to fix some random chan_h323 compilation failures
After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
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r267399 | rmudgett | 2010-06-02 20:02:14 -0400 (Wed, 02 Jun 2010) | 8 lines
Add ETSI Message Waiting Indication (MWI) support.
Add the ability to report waiting messages to ISDN endpoints (phones).
Relevant specification: EN 300 650 and EN 300 745
Review: https://reviewboard.asterisk.org/r/599/
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r267445 | russell | 2010-06-03 10:48:09 -0400 (Thu, 03 Jun 2010) | 2 lines
Comment out a rule that likes to run implicitly unnecessarily, breaking builds
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r267490 | russell | 2010-06-03 13:05:30 -0400 (Thu, 03 Jun 2010) | 2 lines
Remove a line that was killing Asterisk on startup.
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r267492 | mmichelson | 2010-06-03 13:09:11 -0400 (Thu, 03 Jun 2010) | 7 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.
Review: https://reviewboard.asterisk.org/r/683/
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r267537 | russell | 2010-06-03 13:31:41 -0400 (Thu, 03 Jun 2010) | 2 lines
Don't stop Asterisk if chan_usbradio isn't configured.
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r267622 | rmudgett | 2010-06-03 14:38:00 -0400 (Thu, 03 Jun 2010) | 1 line
Make compile again.
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r267624 | lmadsen | 2010-06-03 14:53:24 -0400 (Thu, 03 Jun 2010) | 7 lines
Update UPGRADE.txt and CHANGE for CDR functionality changes.
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
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r267669 | tilghman | 2010-06-03 15:46:42 -0400 (Thu, 03 Jun 2010) | 8 lines
Handle OOM errors more gracefully.
(closes issue #17084)
Reported by: falves11
Patches:
issue17084_162_A.diff uploaded by falves11 (license 374)
Tested by: falves11
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r267714 | russell | 2010-06-03 16:41:24 -0400 (Thu, 03 Jun 2010) | 4 lines
Remove a LOG_WARNING.
This came up when using the sample configs, and just indicates expected behavior.
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r267775 | tilghman | 2010-06-03 21:20:17 -0400 (Thu, 03 Jun 2010) | 14 lines
Merged revisions 267759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r267819 | tilghman | 2010-06-03 21:36:46 -0400 (Thu, 03 Jun 2010) | 2 lines
If there's a default, turn it on, even when the option isn't specified.
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r267862 | tilghman | 2010-06-03 22:58:55 -0400 (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
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r267863 | twilson | 2010-06-03 23:11:31 -0400 (Thu, 03 Jun 2010) | 6 lines
Send an ACK for every final response received for an INVITE
From issue ABE-2247. RFC 3261 compliance for sections 13.2.24 and 17.1.1.2.
Review: https://reviewboard.asterisk.org/r/692/
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r267877 | tilghman | 2010-06-03 23:20:47 -0400 (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
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r267928 | rmudgett | 2010-06-04 10:45:03 -0400 (Fri, 04 Jun 2010) | 15 lines
Incoming overlap dialing no longer works after sig_pri extraction.
The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
(closes issue #17401)
Reported by: avalentin
Patches:
issue17401_digit_timeout.patch uploaded by rmudgett (license 664)
Tested by: avalentin, rmudgett
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r267972 | tilghman | 2010-06-04 12:31:25 -0400 (Fri, 04 Jun 2010) | 9 lines
Merged revisions 267971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010) | 2 lines
As-fixiate the build process
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r268051 | tilghman | 2010-06-04 15:40:00 -0400 (Fri, 04 Jun 2010) | 13 lines
Merged revisions 268050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04 Jun 2010) | 6 lines
Build menuselect with the build environment's compiler, not the host (target)'s compiler.
(closes issue #17464)
Reported by: pprindeville
Tested by: tilghman
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r268127 | tilghman | 2010-06-04 16:42:27 -0400 (Fri, 04 Jun 2010) | 9 lines
Merged revisions 268126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04 Jun 2010) | 2 lines
AC_CONFIG_SUBDIRS has a bad side-effect on cross-compiles.
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r268205 | dvossel | 2010-06-04 17:55:14 -0400 (Fri, 04 Jun 2010) | 27 lines
RFC3261 compliant sip unreliable retransmit timing + 'registerattempts' option tweak
Changes.
1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is 500ms.
Asterisk was starting the retransmission timers at T1*2 which shouldn't
cause any problems, but is not RFC compliant.
2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction,
if the retransmit timer fires while in the proceeding state that
the request must be retransmitted. Asterisk currently ack's
requests for both INVITE and non-INVITE transactions when a
1XX response is received, this patch changes this for non-INVITE requests.
3. The 'registerattempts' option in sip.conf is supposed to set
how many registry attempts will be made before giving up. When
this option is set to 1, I would expect only one registry attempt
to be made before stopping because of a failure, but instead two are
made. In my opinion this is not expected behavior. This option does
not indicate that these are re-attempts. The logic behind this option
has been changed to only attempt registers the exact number of times
this option is set to. If this option is 0, it still continues to
re-attempt the registration forever.
Review: https://reviewboard.asterisk.org/r/687/
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r268281 | dvossel | 2010-06-04 18:37:13 -0400 (Fri, 04 Jun 2010) | 2 lines
fixes compile error from uninitialized variable
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r268321 | tilghman | 2010-06-04 22:51:34 -0400 (Fri, 04 Jun 2010) | 10 lines
Merged revisions 268320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines
Rest In Peace
http://www.outandaboutnewspaper.com/article/4061
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r268395 | kpfleming | 2010-06-05 01:12:34 -0400 (Sat, 05 Jun 2010) | 1 line
Grammatical error fix.
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r268417 | kpfleming | 2010-06-05 01:23:02 -0400 (Sat, 05 Jun 2010) | 1 line
Typo fix.
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r268454 | tilghman | 2010-06-05 13:27:12 -0400 (Sat, 05 Jun 2010) | 5 lines
Verify event is not NULL before attempting to lower its usecount.
(closes issue #17234)
Reported by: mav3rick
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r268456 | tilghman | 2010-06-05 13:55:28 -0400 (Sat, 05 Jun 2010) | 14 lines
Fix crash in DTMF detection.
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
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r268495 | tilghman | 2010-06-05 20:37:30 -0400 (Sat, 05 Jun 2010) | 2 lines
Finally track down and eliminate the "FRACK! warnings from chan_iax2".
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r268534 | tilghman | 2010-06-06 01:29:50 -0400 (Sun, 06 Jun 2010) | 8 lines
Take advantage of variable substitution already in the Makefile to specify the correct location for files in init.d.
(closes issue #16979)
Reported by: jw-asterisk
(issue #15691)
Reported by: itamarjp
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r268578 | rmudgett | 2010-06-07 11:51:39 -0400 (Mon, 07 Jun 2010) | 6 lines
Suppress warning in waitstream_core().
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().
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r268653 | tilghman | 2010-06-07 13:14:40 -0400 (Mon, 07 Jun 2010) | 5 lines
Avoid unloading res_smdi twice.
(closes issue #17237)
Reported by: pabelanger
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r268690 | pabelanger | 2010-06-07 13:34:45 -0400 (Mon, 07 Jun 2010) | 11 lines
Set threshold for silence detection defaults to 256
(closes issue #15685)
Reported by: david_s5
Patches:
dsp-silence-threshold-init.diff uploaded by dant (license 670)
issue15685.patch.v5 uploaded by pabelanger (license 224)
Tested by: danti
Review: https://reviewboard.asterisk.org/r/670/
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r268731 | tilghman | 2010-06-07 14:59:27 -0400 (Mon, 07 Jun 2010) | 4 lines
Event well was going dry.
(issue #17234)
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r268734 | rmudgett | 2010-06-07 15:06:55 -0400 (Mon, 07 Jun 2010) | 2 lines
Moved AOC request code out of the middle of code parsing the dialed number.
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r268773 | tilghman | 2010-06-07 15:52:39 -0400 (Mon, 07 Jun 2010) | 5 lines
Seems strange (and the code backs up) that if the max and min of a statistic is expressed as a double, the last value would not also need to be a double.
(closes issue #15807)
Reported by: klaus3000
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r268774 | rmudgett | 2010-06-07 16:04:42 -0400 (Mon, 07 Jun 2010) | 20 lines
Extract sig_ss7 out of chan_dahdi.
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
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r268817 | tilghman | 2010-06-07 18:47:13 -0400 (Mon, 07 Jun 2010) | 9 lines
Mailbox list would previously grow at each reload, containing duplicates.
Also, optimize the allocation of mailboxes to avoid additional memory structures.
(closes issue #16320)
Reported by: Marquis
Patches:
20100525__issue16320.diff.txt uploaded by tilghman (license 14)
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r268818 | tilghman | 2010-06-07 18:56:53 -0400 (Mon, 07 Jun 2010) | 2 lines
Use the mailbox destructor function, instead.
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r268857 | rmudgett | 2010-06-07 20:45:13 -0400 (Mon, 07 Jun 2010) | 1 line
Make SIP tests compile again.
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r268894 | twilson | 2010-06-08 01:29:08 -0400 (Tue, 08 Jun 2010) | 17 lines
Add SRTP support for Asterisk
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/
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r268896 | tilghman | 2010-06-08 02:16:43 -0400 (Tue, 08 Jun 2010) | 2 lines
Fix trunk build on Mac OS X.
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r268933 | tilghman | 2010-06-08 02:57:24 -0400 (Tue, 08 Jun 2010) | 2 lines
Release list lock before returning on error.
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r268969 | lmadsen | 2010-06-08 10:38:18 -0400 (Tue, 08 Jun 2010) | 7 lines
Fix some doxygen warnings.
(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
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r268988 | lmadsen | 2010-06-08 11:23:20 -0400 (Tue, 08 Jun 2010) | 8 lines
Update note in sip.conf.sample.
Update note in sip.conf.sample about externip and externhost with STUN.
(closes issue #16323)
Reported by: klaus3000
Patches:
sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
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r269007 | seanbright | 2010-06-08 11:39:52 -0400 (Tue, 08 Jun 2010) | 18 lines
Merged revisions 269006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun 2010) | 11 lines
Reduce startup time for cdr_tds with large CDR tables.
Since we are just checking for table existence, add a WHERE clause that will
return no rows but will raise an error if the table doesn't exist.
(closes issue #17380)
Reported by: kkwong
Patches:
issue17380-01.patch uploaded by seanbright (license 71)
Tested by: kkwong
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r269008 | russell | 2010-06-08 11:41:23 -0400 (Tue, 08 Jun 2010) | 5 lines
Ensure CONFIG_FLAGS makes it into the build rules when doing out of tree builds.
(closes issue #16685)
Reported by: pprindeville
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r269083 | mnicholson | 2010-06-08 14:50:45 -0400 (Tue, 08 Jun 2010) | 9 lines
Don't pass null to manager_event()
(closes issue #17087)
Reported by: bklang
Patches:
app-fax-null-sprintf1.diff uploaded by mnicholson (license 96)
Tested by: bklang
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r269119 | tilghman | 2010-06-08 18:45:16 -0400 (Tue, 08 Jun 2010) | 2 lines
Fix build on Mac OS X (and maybe FreeBSD, too)
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r269153 | snuffy | 2010-06-08 19:48:17 -0400 (Tue, 08 Jun 2010) | 11 lines
Add High Resolution Times to CDRs for Asterisk
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
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r269187 | russell | 2010-06-09 06:18:24 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libgtk2.0-dev to the packages list for install_prereq.
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r269196 | russell | 2010-06-09 06:21:23 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libmysqlclient-dev to install_prereq.
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r269197 | russell | 2010-06-09 06:23:05 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libbluetooth-dev to install_prereq.
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r269198 | russell | 2010-06-09 06:28:27 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libradiusclient-ng-dev to install_prereq.
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r269199 | russell | 2010-06-09 06:30:32 -0400 (Wed, 09 Jun 2010) | 2 lines
Add freetds-dev to install_prereq.
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r269200 | russell | 2010-06-09 06:33:32 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libcurl to install_prereq.
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r269201 | russell | 2010-06-09 06:45:10 -0400 (Wed, 09 Jun 2010) | 2 lines
Add an "install-unpackaged" command to install_prereq for installing unpackaged dependencies (such as NBS and libresample).
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r269202 | russell | 2010-06-09 06:47:19 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libopenais-dev to install_prereq.
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r269203 | russell | 2010-06-09 06:48:29 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libnewt-dev to install-prereq.
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r269204 | russell | 2010-06-09 06:53:26 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libpopt-dev, libical-dev, and libspandsp-dev to install_prereq.
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r269205 | russell | 2010-06-09 06:55:07 -0400 (Wed, 09 Jun 2010) | 2 lines
Add libjack-dev to install_prereq.
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r269238 | tzafrir | 2010-06-09 09:17:43 -0400 (Wed, 09 Jun 2010) | 14 lines
dial by name in chan_dahdi
* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
(gGrR) dialing, which make it lsightly more complicated.
https://reviewboard.asterisk.org/r/535/
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r269271 | dvossel | 2010-06-09 11:09:25 -0400 (Wed, 09 Jun 2010) | 15 lines
fixes crash in moh when cachertclasses flag is used
The result for moh_register was not verified to guarantee
the mohclass as added to the container.
(closes issue #16993)
Reported by: dmitri
Patches:
res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
moh_crash2.diff uploaded by dvossel (license 671)
Tested by: dmitri
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r269307 | rmudgett | 2010-06-09 12:54:38 -0400 (Wed, 09 Jun 2010) | 12 lines
Eliminate deadlock potential in dahdi_fixup().
Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are
in a potentially inconsistent state is a potentially bad thing in
principle.
However, calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.
................
r269308 | rmudgett | 2010-06-09 13:06:41 -0400 (Wed, 09 Jun 2010) | 2 lines
Add missing API function to sig_ss7: sig_ss7_fixup().
................
r269346 | pabelanger | 2010-06-09 13:32:52 -0400 (Wed, 09 Jun 2010) | 19 lines
Merged revisions 269334 via svnmerge from
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
........
................
r269417 | russell | 2010-06-09 17:11:43 -0400 (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.
................
r269486 | qwell | 2010-06-09 17:38:33 -0400 (Wed, 09 Jun 2010) | 12 lines
Blocked revisions 269426 via svnmerge
........
r269426 | qwell | 2010-06-09 16:19:17 -0500 (Wed, 09 Jun 2010) | 6 lines
Let systems without a working fork() use res_musiconhold.
Files mode doesn't require anything special, so that can still be used just fine.
AST-357
........
................
r269497 | russell | 2010-06-09 18:19:20 -0400 (Wed, 09 Jun 2010) | 9 lines
Merged revisions 269495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines
Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
........
................
r269569 | russell | 2010-06-09 19:56:08 -0400 (Wed, 09 Jun 2010) | 2 lines
Attempt to fix FreeBSD build problem.
................
r269602 | russell | 2010-06-09 20:32:31 -0400 (Wed, 09 Jun 2010) | 4 lines
Attempt to fix a FreeBSD build error by including sys/stat.h.
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
................
r269636 | tilghman | 2010-06-10 04:15:45 -0400 (Thu, 10 Jun 2010) | 16 lines
Merged revisions 269635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines
Ensure signals are not blocked inside other signal handlers.
This eliminates the annoying <beep> on the console.
(closes issue #17477)
Reported by: jvandal
Patches:
20100610__issue17477.diff.txt uploaded by tilghman (license 14)
........
................
r269707 | kpfleming | 2010-06-10 08:28:17 -0400 (Thu, 10 Jun 2010) | 3 lines
Ensure that 'logger show channels' works properly when wildcards are used in logger.conf.
................
r269711 | russell | 2010-06-10 09:17:51 -0400 (Thu, 10 Jun 2010) | 2 lines
Fix an off by one error that caused a unit test to occasionally crash.
................
r269749 | mmichelson | 2010-06-10 13:14:38 -0400 (Thu, 10 Jun 2010) | 5 lines
Add documentation explaining PLC in Asterisk.
Review: https://reviewboard.asterisk.org/r/688/
................
r269822 | mmichelson | 2010-06-10 15:34:03 -0400 (Thu, 10 Jun 2010) | 25 lines
Merged revisions 269821 via svnmerge from
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.
........
................
r269889 | pabelanger | 2010-06-10 16:30:44 -0400 (Thu, 10 Jun 2010) | 8 lines
Remove ASTBINDIR variable
(closes issue #17031)
Reported by: pabelanger
Patches:
Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman
................
Added:
team/eliel/data_api_providers_gsoc2010/channels/sig_ss7.c
- copied unchanged from r269889, trunk/channels/sig_ss7.c
team/eliel/data_api_providers_gsoc2010/channels/sig_ss7.h
- copied unchanged from r269889, trunk/channels/sig_ss7.h
team/eliel/data_api_providers_gsoc2010/channels/sip/include/sdp_crypto.h
- copied unchanged from r269889, trunk/channels/sip/include/sdp_crypto.h
team/eliel/data_api_providers_gsoc2010/channels/sip/include/srtp.h
- copied unchanged from r269889, trunk/channels/sip/include/srtp.h
team/eliel/data_api_providers_gsoc2010/channels/sip/sdp_crypto.c
- copied unchanged from r269889, trunk/channels/sip/sdp_crypto.c
team/eliel/data_api_providers_gsoc2010/channels/sip/srtp.c
- copied unchanged from r269889, trunk/channels/sip/srtp.c
team/eliel/data_api_providers_gsoc2010/doc/advice_of_charge.txt
- copied unchanged from r269889, trunk/doc/advice_of_charge.txt
team/eliel/data_api_providers_gsoc2010/doc/tex/plc.tex
- copied unchanged from r269889, trunk/doc/tex/plc.tex
team/eliel/data_api_providers_gsoc2010/doc/tex/secure-calls.tex
- copied unchanged from r269889, trunk/doc/tex/secure-calls.tex
team/eliel/data_api_providers_gsoc2010/include/asterisk/aoc.h
- copied unchanged from r269889, trunk/include/asterisk/aoc.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/res_srtp.h
- copied unchanged from r269889, trunk/include/asterisk/res_srtp.h
team/eliel/data_api_providers_gsoc2010/main/aoc.c
- copied unchanged from r269889, trunk/main/aoc.c
team/eliel/data_api_providers_gsoc2010/res/res_srtp.c
- copied unchanged from r269889, trunk/res/res_srtp.c
team/eliel/data_api_providers_gsoc2010/res/res_srtp.exports.in
- copied unchanged from r269889, trunk/res/res_srtp.exports.in
team/eliel/data_api_providers_gsoc2010/tests/test_aoc.c
- copied unchanged from r269889, trunk/tests/test_aoc.c
Modified:
team/eliel/data_api_providers_gsoc2010/ (props changed)
team/eliel/data_api_providers_gsoc2010/CHANGES
team/eliel/data_api_providers_gsoc2010/Makefile
team/eliel/data_api_providers_gsoc2010/Makefile.rules
team/eliel/data_api_providers_gsoc2010/UPGRADE.txt
team/eliel/data_api_providers_gsoc2010/addons/cdr_mysql.c
team/eliel/data_api_providers_gsoc2010/apps/app_dial.c
team/eliel/data_api_providers_gsoc2010/apps/app_fax.c
team/eliel/data_api_providers_gsoc2010/apps/app_meetme.c
team/eliel/data_api_providers_gsoc2010/apps/app_queue.c
team/eliel/data_api_providers_gsoc2010/apps/app_voicemail.c
team/eliel/data_api_providers_gsoc2010/autoconf/ast_ext_lib.m4
team/eliel/data_api_providers_gsoc2010/autoconf/libcurl.m4
team/eliel/data_api_providers_gsoc2010/build_tools/menuselect-deps.in
team/eliel/data_api_providers_gsoc2010/cdr/cdr_adaptive_odbc.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_odbc.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_pgsql.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_sqlite.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_tds.c
team/eliel/data_api_providers_gsoc2010/channels/Makefile
team/eliel/data_api_providers_gsoc2010/channels/chan_dahdi.c
team/eliel/data_api_providers_gsoc2010/channels/chan_iax2.c
team/eliel/data_api_providers_gsoc2010/channels/chan_oss.c
team/eliel/data_api_providers_gsoc2010/channels/chan_sip.c
team/eliel/data_api_providers_gsoc2010/channels/chan_usbradio.c
team/eliel/data_api_providers_gsoc2010/channels/h323/Makefile.in
team/eliel/data_api_providers_gsoc2010/channels/sig_pri.c
team/eliel/data_api_providers_gsoc2010/channels/sig_pri.h
team/eliel/data_api_providers_gsoc2010/channels/sip/dialplan_functions.c
team/eliel/data_api_providers_gsoc2010/channels/sip/include/sip.h
team/eliel/data_api_providers_gsoc2010/codecs/codec_adpcm.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_alaw.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_dahdi.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_g722.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_g726.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_gsm.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_lpc10.c
team/eliel/data_api_providers_gsoc2010/codecs/codec_ulaw.c
team/eliel/data_api_providers_gsoc2010/configs/cdr_custom.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cdr_odbc.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cdr_sqlite3_custom.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cdr_syslog.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cdr_tds.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/chan_dahdi.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/manager.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/sip.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/voicemail.conf.sample
team/eliel/data_api_providers_gsoc2010/configure
team/eliel/data_api_providers_gsoc2010/configure.ac
team/eliel/data_api_providers_gsoc2010/contrib/init.d/rc.debian.asterisk
team/eliel/data_api_providers_gsoc2010/contrib/init.d/rc.redhat.asterisk
team/eliel/data_api_providers_gsoc2010/contrib/scripts/install_prereq
team/eliel/data_api_providers_gsoc2010/doc/tex/asterisk.tex
team/eliel/data_api_providers_gsoc2010/funcs/func_cdr.c
team/eliel/data_api_providers_gsoc2010/funcs/func_channel.c
team/eliel/data_api_providers_gsoc2010/funcs/func_odbc.c
team/eliel/data_api_providers_gsoc2010/include/asterisk/app.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/astmm.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/astobj2.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/autoconfig.h.in
team/eliel/data_api_providers_gsoc2010/include/asterisk/bridging_technology.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/calendar.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/callerid.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/ccss.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/channel.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/data.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/dnsmgr.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/doxyref.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/event.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/features.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/frame.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/global_datastores.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/http.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/localtime.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/logger.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/manager.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/pbx.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/rtp_engine.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/slin.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/smdi.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/threadstorage.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/timing.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/translate.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/xml.h
team/eliel/data_api_providers_gsoc2010/main/Makefile
team/eliel/data_api_providers_gsoc2010/main/ast_expr2.y
team/eliel/data_api_providers_gsoc2010/main/ast_expr2f.c
team/eliel/data_api_providers_gsoc2010/main/asterisk.c
team/eliel/data_api_providers_gsoc2010/main/asterisk.exports.in
team/eliel/data_api_providers_gsoc2010/main/audiohook.c
team/eliel/data_api_providers_gsoc2010/main/ccss.c
team/eliel/data_api_providers_gsoc2010/main/channel.c
team/eliel/data_api_providers_gsoc2010/main/cli.c
team/eliel/data_api_providers_gsoc2010/main/dsp.c
team/eliel/data_api_providers_gsoc2010/main/event.c
team/eliel/data_api_providers_gsoc2010/main/features.c
team/eliel/data_api_providers_gsoc2010/main/file.c
team/eliel/data_api_providers_gsoc2010/main/global_datastores.c
team/eliel/data_api_providers_gsoc2010/main/lock.c
team/eliel/data_api_providers_gsoc2010/main/logger.c
team/eliel/data_api_providers_gsoc2010/main/manager.c
team/eliel/data_api_providers_gsoc2010/main/pbx.c
team/eliel/data_api_providers_gsoc2010/main/rtp_engine.c
team/eliel/data_api_providers_gsoc2010/main/term.c
team/eliel/data_api_providers_gsoc2010/main/translate.c
[... 13787 lines stripped ...]
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