asterisk/asterisk.git
9 years agoAs signed linear audio data is accessed as 16-bit values, certain processors require...
Tilghman Lesher [Fri, 4 Jun 2010 03:20:47 +0000 (03:20 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@267877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSend an ACK for every final response received for an INVITE
Terry Wilson [Fri, 4 Jun 2010 03:11:31 +0000 (03:11 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAs signed linear audio data is accessed as 16-bit values, certain processors require...
Tilghman Lesher [Fri, 4 Jun 2010 02:58:55 +0000 (02:58 +0000)]
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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267862 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoIf there's a default, turn it on, even when the option isn't specified.
Tilghman Lesher [Fri, 4 Jun 2010 01:36:46 +0000 (01:36 +0000)]
If there's a default, turn it on, even when the option isn't specified.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267819 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 267759 via svnmerge from
Tilghman Lesher [Fri, 4 Jun 2010 01:20:17 +0000 (01:20 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@267775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove a LOG_WARNING.
Russell Bryant [Thu, 3 Jun 2010 20:41:24 +0000 (20:41 +0000)]
Remove a LOG_WARNING.

This came up when using the sample configs, and just indicates expected behavior.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoHandle OOM errors more gracefully.
Tilghman Lesher [Thu, 3 Jun 2010 19:46:42 +0000 (19:46 +0000)]
Handle OOM errors more gracefully.

(closes issue #17084)
 Reported by: falves11
 Patches:
       issue17084_162_A.diff uploaded by falves11 (license 374)
 Tested by: falves11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdate UPGRADE.txt and CHANGE for CDR functionality changes.
Leif Madsen [Thu, 3 Jun 2010 18:53:24 +0000 (18:53 +0000)]
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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake compile again.
Richard Mudgett [Thu, 3 Jun 2010 18:38:00 +0000 (18:38 +0000)]
Make compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't stop Asterisk if chan_usbradio isn't configured.
Russell Bryant [Thu, 3 Jun 2010 17:31:41 +0000 (17:31 +0000)]
Don't stop Asterisk if chan_usbradio isn't configured.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove unnecessary code relating to PLC.
Mark Michelson [Thu, 3 Jun 2010 17:09:11 +0000 (17:09 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove a line that was killing Asterisk on startup.
Russell Bryant [Thu, 3 Jun 2010 17:05:30 +0000 (17:05 +0000)]
Remove a line that was killing Asterisk on startup.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoComment out a rule that likes to run implicitly unnecessarily, breaking builds
Russell Bryant [Thu, 3 Jun 2010 14:48:09 +0000 (14:48 +0000)]
Comment out a rule that likes to run implicitly unnecessarily, breaking builds

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267445 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ETSI Message Waiting Indication (MWI) support.
Richard Mudgett [Thu, 3 Jun 2010 00:02:14 +0000 (00:02 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agotry to fix some random chan_h323 compilation failures
Russell Bryant [Wed, 2 Jun 2010 22:46:37 +0000 (22:46 +0000)]
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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ETSI Malicious Call ID support.
Richard Mudgett [Wed, 2 Jun 2010 22:28:58 +0000 (22:28 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix a build error on mac.
Russell Bryant [Wed, 2 Jun 2010 21:44:06 +0000 (21:44 +0000)]
Fix a build error on mac.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267305 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEnsure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been specified.
Russell Bryant [Wed, 2 Jun 2010 21:41:54 +0000 (21:41 +0000)]
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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267303 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ETSI Call Waiting support.
Richard Mudgett [Wed, 2 Jun 2010 21:05:32 +0000 (21:05 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdate CHANGES and aoc help doc to reflect AOC additions
David Vossel [Wed, 2 Jun 2010 19:33:56 +0000 (19:33 +0000)]
Update CHANGES and aoc help doc to reflect AOC additions

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd a CLI command that blocks until Asterisk has fully booted.
Russell Bryant [Wed, 2 Jun 2010 18:53:38 +0000 (18:53 +0000)]
Add a CLI command that blocks until Asterisk has fully booted.

Review: https://reviewboard.asterisk.org/r/684/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267138 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoPrevent use of uninitialized values.
Mark Michelson [Wed, 2 Jun 2010 18:13:18 +0000 (18:13 +0000)]
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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoGeneric Advice of Charge.
Richard Mudgett [Wed, 2 Jun 2010 18:10:15 +0000 (18:10 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSilence a compiler warning.
Russell Bryant [Wed, 2 Jun 2010 17:57:39 +0000 (17:57 +0000)]
Silence a compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267093 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix infinite loop when loading codec speex
Jeff Peeler [Wed, 2 Jun 2010 17:29:35 +0000 (17:29 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@267065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 267009 via svnmerge from
Paul Belanger [Wed, 2 Jun 2010 17:25:05 +0000 (17:25 +0000)]
Merged revisions 267009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  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)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ETSI Advice Of Charge (AOC) event reporting.
Richard Mudgett [Wed, 2 Jun 2010 17:13:53 +0000 (17:13 +0000)]
Add ETSI Advice Of Charge (AOC) event reporting.

This feature generates AMI events in the new aoc event class from the
events passed up by libpri.

Review: https://reviewboard.asterisk.org/r/537/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ETSI Explicit Call Transfer (ECT) support.
Richard Mudgett [Wed, 2 Jun 2010 16:14:12 +0000 (16:14 +0000)]
Add ETSI Explicit Call Transfer (ECT) support.

Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.

Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.

Review: https://reviewboard.asterisk.org/r/520/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agopthread_join to assure the thread is really gone
Paul Belanger [Wed, 2 Jun 2010 13:32:22 +0000 (13:32 +0000)]
pthread_join to assure the thread is really gone

(closes issue #15465)
Reported by: fnordian
Patches:
      bridging.patch uploaded by fnordian (license 110)
Tested by: lmadsen, fnordian, peterh

Review: https://reviewboard.asterisk.org/r/679/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse the correct ical.h file
Terry Wilson [Tue, 1 Jun 2010 22:14:44 +0000 (22:14 +0000)]
Use the correct ical.h file

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSupport setting locale per-mailbox (changes date/time languages for email, pager...
Tilghman Lesher [Tue, 1 Jun 2010 21:28:19 +0000 (21:28 +0000)]
Support setting locale per-mailbox (changes date/time languages for email, pager messages).

(closes issue #14333)
 Reported by: klaus3000
 Patches:
       20090515__issue14333.diff.txt uploaded by tilghman (license 14)
       app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65)
 Tested by: klaus3000

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSet app and appdata fields when a Dial is redirected
Terry Wilson [Tue, 1 Jun 2010 21:12:49 +0000 (21:12 +0000)]
Set app and appdata fields when a Dial is redirected

(closes issue #17204)
Reported by: one47
Tested by: twilson, one47

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266786 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't register functions until the last possible point, so they're not unloaded unnec...
Tilghman Lesher [Tue, 1 Jun 2010 18:02:24 +0000 (18:02 +0000)]
Don't register functions until the last possible point, so they're not unloaded unnecessarily.

(closes issue #15996)
 Reported by: junky
 Patches:
       sdmi_wait.diff uploaded by junky (license 177)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEliminate stale manager events after a set interval, even if AMI clients don't query...
Tilghman Lesher [Tue, 1 Jun 2010 16:41:00 +0000 (16:41 +0000)]
Eliminate stale manager events after a set interval, even if AMI clients don't query for them.

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

9 years agoMerged revisions 266585 via svnmerge from
Tilghman Lesher [Tue, 1 Jun 2010 15:18:59 +0000 (15:18 +0000)]
Merged revisions 266585 via svnmerge from
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

9 years agoNeeds to be wrapped in <para>
Tilghman Lesher [Sun, 30 May 2010 20:18:03 +0000 (20:18 +0000)]
Needs to be wrapped in <para>

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 266437 via svnmerge from
Tilghman Lesher [Sun, 30 May 2010 04:44:28 +0000 (04:44 +0000)]
Merged revisions 266437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) | 2 lines

  Reverting patch and reopening issue #16784, as patch breaks color display.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266438 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix ical library handling (again)
Terry Wilson [Fri, 28 May 2010 22:54:03 +0000 (22:54 +0000)]
Fix ical library handling (again)

Newer versions of libical (which we require) store the header file in a
libical/ subfolder and include an ical.h file that does a #warning for
deprecation and then #includes <libical/ical.h>. Since we now test for
libical/ical.h, we can change the #includes back to <libical/ical.h> and
remove the test which specifically adds /usr/include/libical as an include
directory.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSetup environment variables for the benefit of child processes and disallow changing...
Tilghman Lesher [Fri, 28 May 2010 22:50:06 +0000 (22:50 +0000)]
Setup environment variables for the benefit of child processes and disallow changing them.

(closes issue #14899)
 Reported by: jmls
 Patches:
       20090916__issue14899.diff.txt uploaded by tilghman (license 14)
 Tested by: jmls

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOnly report swap on platforms which can examine those statistics
Tilghman Lesher [Fri, 28 May 2010 20:53:04 +0000 (20:53 +0000)]
Only report swap on platforms which can examine those statistics

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266337 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes crash when creation of UDPTL fails
David Vossel [Fri, 28 May 2010 17:55:38 +0000 (17:55 +0000)]
fixes crash when creation of UDPTL fails

(closes issue #17264)
Reported by: falves11
Patches:
      issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671)
      issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671)
Tested by: falves11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMore build fixes for ical/neon and res_calendar_ews
Terry Wilson [Fri, 28 May 2010 17:34:00 +0000 (17:34 +0000)]
More build fixes for ical/neon and res_calendar_ews

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofix compile error
Jeff Peeler [Thu, 27 May 2010 20:08:49 +0000 (20:08 +0000)]
fix compile error

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266240 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCache query results for one second.
Tilghman Lesher [Thu, 27 May 2010 19:25:16 +0000 (19:25 +0000)]
Cache query results for one second.

Queries from the PBX core come in 3's.  Caching avoids the additional
performance penalty from those two additional queries hitting the database.

(closes issue #16521)
 Reported by: tilghman
 Patches:
       20091229__issue16521.diff.txt uploaded by tilghman (license 14)
 Tested by: Hubguru, tilghman

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 266142 via svnmerge from
Tilghman Lesher [Wed, 26 May 2010 21:17:46 +0000 (21:17 +0000)]
Merged revisions 266142 via svnmerge from
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

9 years agoRemove redundant ast_conntected_line_free call.
Mark Michelson [Wed, 26 May 2010 20:17:54 +0000 (20:17 +0000)]
Remove redundant ast_conntected_line_free call.

This wouldn't cause any problems, but it's certainly not needed either.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266098 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove unrelated MOH change from previous commit.
Mark Michelson [Wed, 26 May 2010 20:15:47 +0000 (20:15 +0000)]
Remove unrelated MOH change from previous commit.

Thanks Kevin!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266094 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix misspelling of macro args.
Mark Michelson [Wed, 26 May 2010 20:04:51 +0000 (20:04 +0000)]
Fix misspelling of macro args.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266092 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agodo all sip registry parsing before transmit_register
David Vossel [Wed, 26 May 2010 19:46:49 +0000 (19:46 +0000)]
do all sip registry parsing before transmit_register

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

9 years agofixes failed SIP Directed pickup resulting in dead channel
David Vossel [Wed, 26 May 2010 18:32:51 +0000 (18:32 +0000)]
fixes failed SIP Directed pickup resulting in dead channel

(closes issue #17339)
Reported by: one47
Patches:
      sip_magic_pickup2 uploaded by one47 (license 23)
Tested by: one47, dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 266004 via svnmerge
Mark Michelson [Wed, 26 May 2010 18:21:53 +0000 (18:21 +0000)]
Blocked revisions 266004 via svnmerge

........
  r266004 | mmichelson | 2010-05-26 13:21:10 -0500 (Wed, 26 May 2010) | 12 lines

  Make AgentComplete message more consistent.

  At times, the "Member" field was not specified during the event.
  It's there now.

  (closes issue #15638)
  Reported by: elbriga
  Patches:
        patchAppQueueAgentComplete.diff uploaded by elbriga (license 482)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266005 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 265910 via svnmerge from
Tilghman Lesher [Wed, 26 May 2010 16:23:28 +0000 (16:23 +0000)]
Merged revisions 265910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines

  Not finding rows in the DB does not rise to the level of a warning.

  (closes issue #17062)
   Reported by: drookie
   Patches:
         20100525__issue17062.diff.txt uploaded by tilghman (license 14)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265923 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoConstruct socket name, according to the Postgres docs, and document as such.
Tilghman Lesher [Wed, 26 May 2010 16:14:48 +0000 (16:14 +0000)]
Construct socket name, according to the Postgres docs, and document as such.

(closes issue #17392)
 Reported by: dps
 Patches:
       20100525__issue17392.diff.txt uploaded by tilghman (license 14)
 Tested by: dps

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265894 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years ago.......
Mark Michelson [Wed, 26 May 2010 14:45:47 +0000 (14:45 +0000)]
.......

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRe-enable "always" option for videosupport option in sip.conf.
Mark Michelson [Wed, 26 May 2010 14:41:55 +0000 (14:41 +0000)]
Re-enable "always" option for videosupport option in sip.conf.

(closes issue #17016)
Reported by: twilson
Patches:
      17016.patch uploaded by mmichelson (license 60)
  Tested by: devmod

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265842 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoEnsure that libneon > 0.29.0 is installed for res_calendar_ews
Terry Wilson [Wed, 26 May 2010 05:33:11 +0000 (05:33 +0000)]
Ensure that libneon > 0.29.0 is installed for res_calendar_ews

This uses a modified version of pabelanger's patch that checks for NTLM support
instead, which was added in 0.29.0 which is what is required for
res_calendar_ews.

(closes issue #17391)
Reported by: loloski
Patches:
      issue17391.patch.v2 uploaded by pabelanger (license 224)
Tested by: twilson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse configure to determine the prefixes and include directories properly.
Tilghman Lesher [Wed, 26 May 2010 00:29:40 +0000 (00:29 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoProperly use peer's outboundproxy for outbound REGISTERs.
Mark Michelson [Tue, 25 May 2010 20:59:04 +0000 (20:59 +0000)]
Properly use peer's outboundproxy for outbound REGISTERs.

The logic used in transmit_register to get the outboundproxy for a peer
was flawed since this value would be overridden shortly afterwards when
create_addr was called.

In addition, this also fixes some logic used when parsing users.conf so
that the peer name is placed in the internally-generated register string
so that an outboundproxy set in the Asterisk GUI will be used for outbound
REGISTERs.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265698 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 265613 via svnmerge
David Vossel [Tue, 25 May 2010 17:12:28 +0000 (17:12 +0000)]
Blocked revisions 265613 via svnmerge

........
  r265613 | dvossel | 2010-05-25 12:11:40 -0500 (Tue, 25 May 2010) | 8 lines

  fixes build issue with zaptel

  (closes issue #17394)
  Reported by: aragon
  Patches:
        half_buffer_fix.diff uploaded by dvossel (license 671)
  Tested by: aragon
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 265610 via svnmerge from
Matthew Nicholson [Tue, 25 May 2010 17:00:11 +0000 (17:00 +0000)]
Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines

  Don't mark the cdr records of unanswered queue calls with "NOANSWER".  This restores the behavior prior to r258670.

  (closes issue #17334)
  Reported by: jvandal
  Patches:
        queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
  Tested by: aragon, jvandal
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265611 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMemory leak in connected line data when SIP blond transfer done.
Richard Mudgett [Tue, 25 May 2010 16:23:51 +0000 (16:23 +0000)]
Memory leak in connected line data when SIP blond transfer done.

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

9 years agoTypos: 'succesful' (lintian)
Tzafrir Cohen [Tue, 25 May 2010 08:31:51 +0000 (08:31 +0000)]
Typos: 'succesful' (lintian)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerge the rest of the FullyBooted patch
Terry Wilson [Mon, 24 May 2010 22:21:58 +0000 (22:21 +0000)]
Merge the rest of the FullyBooted patch

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow SendDTMF to play digits to a specified channel.
Mark Michelson [Mon, 24 May 2010 22:16:29 +0000 (22:16 +0000)]
Allow SendDTMF to play digits to a specified channel.

Patch supplied by reporter was modified to use autoservice and
prevent a potential channel ref leak but is otherwise as the
reporter uploaded it.

(closes issue #17182)
Reported by: rcasas
Patches:
      app_senddtmf.c.patch_trunk uploaded by rcasas (license 641)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265453 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoPrint openh323 log to the Asterisk console.
Mark Michelson [Mon, 24 May 2010 22:05:15 +0000 (22:05 +0000)]
Print openh323 log to the Asterisk console.

(closes issue #17109)
Reported by: under
Patches:
      logstream.diff uploaded by under (license 914)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow type=user SIP endpoints to be loaded properly from realtime.
Mark Michelson [Mon, 24 May 2010 21:44:30 +0000 (21:44 +0000)]
Allow type=user SIP endpoints to be loaded properly from realtime.

(closes issue #16021)
Reported by: Guggemand
Patches:
      realtime-type-fix.patch uploaded by Guggemand (license 897)
  (altered by me slightly to avoid ref leaks)
Tested by: Guggemand

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265449 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake app_rpt.c able to compile again.
Richard Mudgett [Mon, 24 May 2010 20:08:35 +0000 (20:08 +0000)]
Make app_rpt.c able to compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265367 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoreverses incorrect logic introduced by r243200
David Vossel [Mon, 24 May 2010 19:42:54 +0000 (19:42 +0000)]
reverses incorrect logic introduced by r243200

The decoding of the replace_id did not need to be broken
up in this instance.  This was brought to my attention
again because it caused a segfault when the from or to
tags were not present in the "Replaces" header.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265366 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd the FullyBooted AMI event
Terry Wilson [Mon, 24 May 2010 19:06:40 +0000 (19:06 +0000)]
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCalendaring support for Exchange Server 2007+ via EWS
Terry Wilson [Mon, 24 May 2010 18:21:20 +0000 (18:21 +0000)]
Calendaring support for Exchange Server 2007+ via EWS

This commit adds support for calendaring with Exchange Server 2007+ via
Exchange Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature.

(closes issue #17022)
Reported by: pitel
Patches:
      res_calendar_ews.c uploaded by pitel (license 1008)
Tested by: pitel, twilson

Review: https://reviewboard.asterisk.org/r/557/
Review: https://reviewboard.asterisk.org/r/668/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOn systems with a LOT of RAM, a signed integer sometimes printed negative.
Tilghman Lesher [Mon, 24 May 2010 18:19:08 +0000 (18:19 +0000)]
On systems with a LOT of RAM, a signed integer sometimes printed negative.

(closes issue #16837)
 Reported by: jlpedrosa
 Patches:
       20100504__issue16837.diff.txt uploaded by tilghman (license 14)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes segfault when using generic plc
David Vossel [Mon, 24 May 2010 16:10:09 +0000 (16:10 +0000)]
fixes segfault when using generic plc

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agosmall changes to avoiding 'freeing unused memory...'
Alexandr Anikin [Sun, 23 May 2010 18:23:38 +0000 (18:23 +0000)]
small changes to avoiding 'freeing unused memory...'

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoChannel initialization failure causes crashes.
Richard Mudgett [Fri, 21 May 2010 22:46:52 +0000 (22:46 +0000)]
Channel initialization failure causes crashes.

__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

9 years agoMerged revisions 265089 via svnmerge from
Mark Michelson [Fri, 21 May 2010 21:08:51 +0000 (21:08 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBe sure to set the sin_family on the proxy when allocating.
Mark Michelson [Fri, 21 May 2010 20:38:14 +0000 (20:38 +0000)]
Be sure to set the sin_family on the proxy when allocating.

(closes issue #17157)
Reported by: stuarth

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 264999 via svnmerge from
Mark Michelson [Fri, 21 May 2010 16:54:21 +0000 (16:54 +0000)]
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.
........

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9 years agoMerged revisions 264996 via svnmerge from
Mark Michelson [Fri, 21 May 2010 16:44:27 +0000 (16:44 +0000)]
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/
........

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9 years agoLog spandsp's fax debug output to the FAX logger level.
Mark Michelson [Fri, 21 May 2010 15:15:58 +0000 (15:15 +0000)]
Log spandsp's fax debug output to the FAX logger level.

Review: https://reviewboard.asterisk.org/r/658

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9 years agoTake dup'd code for directmedia ACLs and make utility func
Terry Wilson [Fri, 21 May 2010 01:00:44 +0000 (01:00 +0000)]
Take dup'd code for directmedia ACLs and make utility func

The same code was repeated in lots of different places, so I made a utility
fuction for it. This should make the merge in the v6-new branch easier.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 264820 via svnmerge from
Richard Mudgett [Thu, 20 May 2010 23:29:43 +0000 (23:29 +0000)]
Merged revisions 264820 via svnmerge from
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.
........

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9 years agoLet ExtensionState resolve dynamic hints.
Tilghman Lesher [Thu, 20 May 2010 22:23:32 +0000 (22:23 +0000)]
Let ExtensionState resolve dynamic hints.

(closes issue #16623)
 Reported by: tilghman
 Patches:
       20100116__issue16623.diff.txt uploaded by tilghman (license 14)
 Tested by: lmadsen

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9 years agoError message fix.
Tilghman Lesher [Thu, 20 May 2010 21:28:53 +0000 (21:28 +0000)]
Error message fix.

(closes issue #17356)
 Reported by: kenner
 Patches:
       app_stack.c.diff uploaded by kenner (license 1040)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264752 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAvoid crash in generic CC agent init if caller name or number is NULL.
Richard Mudgett [Thu, 20 May 2010 20:49:40 +0000 (20:49 +0000)]
Avoid crash in generic CC agent init if caller name or number is NULL.

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9 years agoDial and queue connected line update macro not always run when expected.
Richard Mudgett [Thu, 20 May 2010 19:40:03 +0000 (19:40 +0000)]
Dial and queue connected line update macro not always run when expected.

The connected line update macro would not get run if the connected line
number string was empty.  The number could be empty if the connected line
update did not update a number but the name.  It should be run if there
was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and
queues.

Renamed and added some more comments for some confusing identifiers
directly connected to the related code.

Also fixed a memory leak in app_queue.

Review: https://reviewboard.asterisk.org/r/669/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264669 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd support for direct media ACLs
Terry Wilson [Thu, 20 May 2010 17:54:02 +0000 (17:54 +0000)]
Add support for direct media ACLs

directmediapermit/directmediadeny support to restrict which peers can do
directmedia based on ip address. In some networks not all phones are fully
routed, i.e. not all phones can ping each other. This patch adds a way to
restrict directmedia for certain peers between certain networks.

(closes issue #16645)
Reported by: raarts
Patches:
      directmediapermit.patch uploaded by raarts (license 937)
Tested by: raarts

Review: https://reviewboard.asterisk.org/r/467/

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9 years agoIgnore pre-processed source files generated during DONT_OPTIMIZE dev-mode builds.
Kevin P. Fleming [Thu, 20 May 2010 15:30:19 +0000 (15:30 +0000)]
Ignore pre-processed source files generated during DONT_OPTIMIZE dev-mode builds.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCorrect 'all logger levels' patch to work properly.
Kevin P. Fleming [Thu, 20 May 2010 12:06:11 +0000 (12:06 +0000)]
Correct 'all logger levels' patch to work properly.

Nick Lewis pointed out that the patch as committed wouldn't actually include
dynamic logger levels, which was missed by the other reviewers. Thanks!

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9 years agoFix transcode_via_sln option with SIP calls and improve PLC usage.
Mark Michelson [Wed, 19 May 2010 21:29:08 +0000 (21:29 +0000)]
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: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofixes infinite loop during udptl.c's decode_open_type
David Vossel [Wed, 19 May 2010 20:30:33 +0000 (20:30 +0000)]
fixes infinite loop during udptl.c's decode_open_type

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)

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9 years agoCast an unsigned int to a signed int when comparing it with 0.
Matthew Nicholson [Wed, 19 May 2010 20:26:27 +0000 (20:26 +0000)]
Cast an unsigned int to a signed int when comparing it with 0.

(AST-377)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264379 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 264334 via svnmerge from
Matthew Nicholson [Wed, 19 May 2010 20:02:57 +0000 (20:02 +0000)]
Merged revisions 264334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines

  Set quieted flag when receiving a dtmf tone during playback in speechbackground.

  (closes issue #16966)
  Reported by: asackheim
........

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9 years agofixes crash in check_rtp_timeout
David Vossel [Wed, 19 May 2010 19:21:04 +0000 (19:21 +0000)]
fixes crash in check_rtp_timeout

During deadlock avoidance the sip dialog pvt is locked and
unlocked.  When this occurs we have no guarantee the pvt's owner
is still valid.  We were trying to access the pvt's owner after
this without checking to see if it still existed first.

(closes issue #17271)
Reported by: under
Patches:
      check_rtp_timeout.diff uploaded by under (license 914)
Tested by: dvossel

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 264248 via svnmerge from
Tilghman Lesher [Wed, 19 May 2010 17:48:31 +0000 (17:48 +0000)]
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
........

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9 years agoKeep track of digit duration, when we're decoding inband to pass DTMF frames.
Tilghman Lesher [Wed, 19 May 2010 16:42:20 +0000 (16:42 +0000)]
Keep track of digit duration, when we're decoding inband to pass DTMF frames.

(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

9 years agoFix compilation problem with previous commit.
Leif Madsen [Wed, 19 May 2010 15:39:39 +0000 (15:39 +0000)]
Fix compilation problem with previous commit.

(issue #16009)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ability for logger channels to include *all* levels.
Kevin P. Fleming [Wed, 19 May 2010 15:29:28 +0000 (15:29 +0000)]
Add ability for logger channels to include *all* levels.

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/

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9 years agoAdd ability to hangup all channels from the CLI.
Leif Madsen [Wed, 19 May 2010 15:12:18 +0000 (15:12 +0000)]
Add ability to hangup all channels from the CLI.

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

9 years agofixes crash during dtmf
David Vossel [Wed, 19 May 2010 14:38:02 +0000 (14:38 +0000)]
fixes crash during dtmf

During the processing of Cisco dtmf the dtmf samples were
not being calculated correctly.  In an attempt to determine
what sample rate was being used, a NULL frame was processed
which caused a crash.  This patch resolves this.

(closes issue #17248)
Reported by: falves11
Patches:
      issue_17248.diff uploaded by dvossel (license 671)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264114 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofix incorrectly typed indications for [nz] stutter and dialrecall
Alec L Davis [Wed, 19 May 2010 08:09:14 +0000 (08:09 +0000)]
fix incorrectly typed indications for [nz] stutter and dialrecall

(closes issue #17359)
Reported by: alecdavis
Patches:
      bug17359.diff.txt uploaded by alecdavis (license 585)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264031 65c4cc65-6c06-0410-ace0-fbb531ad65f3