[asterisk-commits] phsultan: branch phsultan/jabberreceive r188899 - in /team/phsultan/jabberrec...

[SVN commits to the Asterisk project]

Author: phsultan
Date: Fri Apr 17 07:39:54 2009
New Revision: 188899

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=188899
Log:
Merged revisions 177320,177356,177384,177387,177451,177506,177537,177595,177624,177661,177664,177697,177699,177732,177787,177849,177852,177855,177884,177913,177944,177988,178022,178027,178030,178061,178107,178142,178213,178300,178303,178342,178374-178375,178381,178446,178509,178573,178605,178607,178641,178703-178704,178733,178764,178767,178801,178828,178848,178870-178871,178919,178986,179021,179057,179122,179154,179161,179164,179219,179254,179291,179323,179361,179396,179462,179465,179469,179533,179537,179609,179672,179675,179742,179745,179841,179903,179937,179972-179973,180007,180011,180032,180079,180120,180155,180195,180259,180261,180304,180334,180369,180373,180382-180383,180465,180534,180579,180641,180684,180719,180750,180800,180859,180862,180898,180935,180938,180942,180944,181027-181028,181032-181033,181099,181134-181135,181210,181244,181292,181296,181301,181345,181371,181424,181428,181444,181465,181499,181542,181577,181612,181656,181661,181665,181731,181769,181846,181899,181985,182022,182029,182071,182121,182171,182211,182278,182282,182355,182362,182408,182450,182521,182525,182530,182553,182596,182607,182653,182722,182762,182826,182847-182848,182883,182960,182964,182966,183028,183032,183057,183108,183117,183124,183148,183172,183196,183239,183242,183244,183312,183321,183345,183436,183511,183553-183555,183560,183652,183701,183766,183831,183865,183914,183995,184037,184043,184079,184147,184151,184219-184220,184280,184339,184344,184389,184448,184512,184515,184531,184566,184628,184630,184639,184673,184677,184693,184726,184762,184798,184801,184838,184843,184910,184948,184986,185072,185122-185123,185197,185261,185299,185363,185432,185469,185532,185581,185600,185604,185664,185704,185741,185772,185777,185846,185912,185953,186021,186058,186060,186078,186101,186175,186230,186286,186297,186321,186379,186382,186444,186447,186461,186525,186537,186563,186566,186620,186624,186653,186687,186720,186799,186833,186837,186842,186899,186928,186953,186957,186985,187036,187046,187050,187105,187108,187138,187179,187210-187211,187269,187302,187360-187361,187363,187381,187421,187424,187426,187483,187488,187491,187556,187560,187599,187634-187636,187673-187675,187680,187714,187721,187764,187770,187772-187773,187830,187866,187906,187963,188032,188067,188102,188150,188206,188210,188247,188283-188284,188342,188378,188413,188470,188515,188544,188585,188647,188705,188742,188774,188836 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r177320 | tilghman | 2009-02-19 01:26:01 +0100 (Thu, 19 Feb 2009) | 2 lines

ODBC transaction support

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r177356 | jpeeler | 2009-02-19 16:56:31 +0100 (Thu, 19 Feb 2009) | 4 lines

Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the
asterisk-dev mailing list. Thanks!


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r177384 | file | 2009-02-19 17:38:41 +0100 (Thu, 19 Feb 2009) | 10 lines

Merged revisions 177383 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
  
  If we are able to create a speech structure unset the ERROR variable in case it was previously set.
  (issue #LUMENVOX-13)
........

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r177387 | jpeeler | 2009-02-19 17:45:02 +0100 (Thu, 19 Feb 2009) | 3 lines

Fix another merge error from 176708


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r177451 | oej | 2009-02-19 20:14:05 +0100 (Thu, 19 Feb 2009) | 13 lines

Blocking MWI change to 1.4 since the mwi system is different in 1.6.x and trunk. Will do some
testing to make sure this works properly, but from reading the code, it does seem to work
as it should.

Blocked revisions 177450 via svnmerge

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r177450 | oej | 2009-02-19 19:58:57 +0100 (Tor, 19 Feb 2009) | 2 lines

Force a MWI notification after subscribe request. Reported by the Resiprocate dev team. Thanks!

........

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r177506 | tilghman | 2009-02-19 20:46:13 +0100 (Thu, 19 Feb 2009) | 2 lines

Document how to use database transactions

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r177537 | tilghman | 2009-02-19 23:33:00 +0100 (Thu, 19 Feb 2009) | 14 lines

Merged revisions 177536 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines
  
  Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
  (closes issue #14253)
   Reported by: Skavin
   Patches: 
         20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Skavin
........

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r177595 | murf | 2009-02-20 00:56:50 +0100 (Fri, 20 Feb 2009) | 32 lines

Merged revisions 177540 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.

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  r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
  
  This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
  
  The real culprit was the --full argument to flex
  in the Makefile! This causes a 7-bit scanner to be
  generated.
  
  I reviewed the rules and found one rule where I needed
  to specifically include 8-bit chars for a token.
  
  I tested against the text supplied by ibercom, and 
  all looks very well.
  
  This has been there a surprisingly long time!
  
  
  (closes issue #14498)
  Reported by: ibercom
  Patches:
        14498.patch uploaded by murf (license 17)
  Tested by: murf
........

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r177624 | jpeeler | 2009-02-20 01:35:53 +0100 (Fri, 20 Feb 2009) | 7 lines

Set sip_request ast_str data to NULL so ast_str_copy allocates space properly
in copy_request

(issue #14478)
Reported by: erik_dedecker


................
r177661 | tilghman | 2009-02-20 18:22:19 +0100 (Fri, 20 Feb 2009) | 2 lines

Oops, merge broke trunk

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r177664 | tilghman | 2009-02-20 18:29:51 +0100 (Fri, 20 Feb 2009) | 8 lines

Allow semicolons to be escaped, when passing arguments to the System command.
(closes issue #14231)
 Reported by: jcovert
 Patches: 
       20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
       corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
 Tested by: jcovert

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r177697 | dvossel | 2009-02-20 21:18:40 +0100 (Fri, 20 Feb 2009) | 13 lines

Blocked revisions 177696 via svnmerge

........
  r177696 | dvossel | 2009-02-20 14:17:37 -0600 (Fri, 20 Feb 2009) | 8 lines
  
  Fixes issue with undefined audio codecs in chan_iax2
  
  During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec.  In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined.  By default all bits are enabled unless specified otherwise.  Since its a 2 byte field and 13-15 are not defined, these bits are never turned off.  In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk.  I fixed this by adding #define for undefined audio codec bits.  These bits are then removed from iax2's full bandwidth capabilities.   
  
  (closes issue #14283)
  Reported by: jcovert
........

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r177699 | dhubbard | 2009-02-20 21:29:00 +0100 (Fri, 20 Feb 2009) | 9 lines

Make app_fax compatible with spandsp-0.0.6pre4

Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session.  The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx.  This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.

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r177732 | tilghman | 2009-02-20 22:25:37 +0100 (Fri, 20 Feb 2009) | 10 lines

Merged revisions 177701 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines
  
  This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
  Fixed for snuff-home on -dev channel.
........

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r177787 | tilghman | 2009-02-21 00:02:35 +0100 (Sat, 21 Feb 2009) | 16 lines

Merged revisions 177786 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
  
  Don't print the CR-NL combination when we aren't outputting to the manager.
  
  An embedded CR-NL in a CLI command screws up several AMI parsers that don't
  expect to see that combination in the middle of output.
  
  (Closes issue #14305)
  Reported by: martins
  Patch by: tilghman
........

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r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines

make chan_sip.c compile on OpenBSD again.

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r177852 | mvanbaak | 2009-02-21 14:13:35 +0100 (Sat, 21 Feb 2009) | 18 lines

set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path

When running asterisk as non-root and without this patch the pidfile wants
to go into /var/run/asterisk.pid. This directory is not writable for
the non-root user and changing permissions is not an option.
Putting it in /var/run/asterisk/asterisk.pid makes it possible
to set permissions on the /var/run/asterisk dir so everything
works as it should be.

Patched committed is based on pabelanger's patch.

(closes issue #13153)
Reported by: pabelanger
Patches:
      2009012900_bug13153-nonrootscripts.diff.txt uploaded by mvanbaak (license 7)

Review: http://reviewboard.digium.com/r/139/

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r177855 | russell | 2009-02-21 14:17:47 +0100 (Sat, 21 Feb 2009) | 5 lines

Fix build issues on Solaris and OpenBSD.

(closes issue #14512)
Reported by: snuffy

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r177884 | seanbright | 2009-02-21 15:16:44 +0100 (Sat, 21 Feb 2009) | 3 lines

Trailing whitespace, minor coding guideline fixes, and start beefing up the
hashtab documentation a bit.

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r177913 | mvanbaak | 2009-02-21 15:37:04 +0100 (Sat, 21 Feb 2009) | 7 lines

add extra check for sysinfo/sysctl

(closes issue #14513)
Reported by: snuffy
Patches:
      bug14513_fixsysinfo.diff uploaded by snuffy (license 35)

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r177944 | tilghman | 2009-02-21 16:59:49 +0100 (Sat, 21 Feb 2009) | 2 lines

On update, test against the existence of sipregs.

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r177988 | mvanbaak | 2009-02-23 00:04:37 +0100 (Mon, 23 Feb 2009) | 13 lines

Add a couple of manager commands to chan_skinny

Added:
SKINNYdevices
SKINNYshowdevice
SKINNYlines
SKINNYshowline

(closes issue #14521)
Reported by: mvanbaak

Review: http://reviewboard.digium.com/r/170/

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r178022 | russell | 2009-02-23 18:29:16 +0100 (Mon, 23 Feb 2009) | 6 lines

Fix a regression in scheduler entry ordering, and add a regression test for it.

(closes issue #14522)
Reported by: pj
Tested by: russell

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r178027 | mvanbaak | 2009-02-23 18:48:32 +0100 (Mon, 23 Feb 2009) | 2 lines

list the addition of the SKINNY manager actions in the CHANGES file.

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r178030 | dvossel | 2009-02-23 18:59:55 +0100 (Mon, 23 Feb 2009) | 7 lines

Changes the way keyrotation is enabled by default

Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE.  the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers".  Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. 

(closes issue #14523)
Reported by: mvanbaak

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r178061 | mvanbaak | 2009-02-23 19:23:38 +0100 (Mon, 23 Feb 2009) | 3 lines

update the new manager commands in chan_skinny to match
chan_sip's headers. requested by oej.

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r178107 | tilghman | 2009-02-23 22:02:18 +0100 (Mon, 23 Feb 2009) | 7 lines

Permit emailsubject and emailbody to be set per mailbox.
(closes issue #14372)
 Reported by: fhackenberger
 Patches: 
       voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592)
       with additional fixes by Corydon76 (license 14)

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r178142 | russell | 2009-02-24 00:11:37 +0100 (Tue, 24 Feb 2009) | 22 lines

Merged revisions 178141 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines

Fix infinite DTMF when a BEGIN is received without an END.

This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem.  The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.

In passing, I removed the dtmfsamples variable which was completed unused.  I
also removed a redundant setting of the lastrxts variable.

(closes issue #14460)
Reported by: moliveras

........

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r178213 | file | 2009-02-24 16:18:38 +0100 (Tue, 24 Feb 2009) | 16 lines

Merged revisions 178205 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
  
  Skip check for extension when subscribing for MWI.
  
  Since the remote side is not actually subscribing to a specific extension when
  subscribing for MWI just skip the check to see if the extension exists. They can't use it
  to specify the mailbox either since we require configuration of that in sip.conf
  
  (closes issue #14531)
  Reported by: festr
........

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r178300 | dvossel | 2009-02-24 18:42:37 +0100 (Tue, 24 Feb 2009) | 12 lines

Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well. 

Manager command "iaxpeers" now shows if a link is trunked and encrypted.  Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.  

(closes issue #14427)
Reported by: snuffy
Patches:
	iax_show_trunks.diff uploaded by snuffy (license 35)
	2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/

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r178303 | tilghman | 2009-02-24 18:51:36 +0100 (Tue, 24 Feb 2009) | 7 lines

Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary.
Also, add some documentation supporting the use of astcanary.
(closes issue #14538)
 Reported by: KNK
 Patches: 
       asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)

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r178342 | tilghman | 2009-02-24 21:06:48 +0100 (Tue, 24 Feb 2009) | 2 lines

Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init.

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r178374 | russell | 2009-02-24 21:39:57 +0100 (Tue, 24 Feb 2009) | 14 lines

Merged revisions 178373 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines

Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly.

(issue #14460)
Reported by: moliveras
Tested by: russell

........

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r178375 | tilghman | 2009-02-24 21:40:02 +0100 (Tue, 24 Feb 2009) | 2 lines

The 3 possible errors with pipe(2) are all impossible in this situation.

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r178381 | tilghman | 2009-02-24 21:52:44 +0100 (Tue, 24 Feb 2009) | 2 lines

Apparently, a void cast doesn't override warn_unused_result.

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r178446 | tilghman | 2009-02-25 00:27:23 +0100 (Wed, 25 Feb 2009) | 12 lines

Merged revisions 178445 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines
  
  Add section about the #exec command in configuration files.
  (closes issue #14540)
   Reported by: jtodd
   Patch by: jtodd, with additional notes by tilghman (license 14) 
........

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r178509 | russell | 2009-02-25 13:45:30 +0100 (Wed, 25 Feb 2009) | 10 lines

Merged revisions 178508 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines

Update the copyright year for the main page of the doxygen documentation.

........

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r178573 | tilghman | 2009-02-25 20:03:35 +0100 (Wed, 25 Feb 2009) | 2 lines

Oops, wrong direction of command

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r178605 | tilghman | 2009-02-25 20:24:44 +0100 (Wed, 25 Feb 2009) | 9 lines

Use notification when timezone files change and re-scan then.
(closes issue #14300)
 Reported by: jamessan
 Patches: 
       20090127__bug14300.diff.txt uploaded by tilghman (license 14)
       20090224__bug14300.diff uploaded by jamessan (license 246)
 Tested by: jamessan
 Review: http://reviewboard.digium.com/r/136/

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r178607 | tilghman | 2009-02-25 20:49:46 +0100 (Wed, 25 Feb 2009) | 2 lines

Picky, picky buildbots

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r178641 | murf | 2009-02-25 22:09:27 +0100 (Wed, 25 Feb 2009) | 22 lines

Blocked revisions 178640 via svnmerge

........
  r178640 | murf | 2009-02-25 14:00:50 -0700 (Wed, 25 Feb 2009) | 17 lines
  
  This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent
  
  While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught
  in tokens; and in so doing, it absorbs the ${ sequence and messes up the
  checking of raw exprs by AEL.
  
  Trunk already has these changes.
  
  
  
  (closes issue #14543)
  Reported by: klaus3000
  Patches:
        patch.14543 uploaded by murf (license 17)
  Tested by: murf
........

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r178703 | kpfleming | 2009-02-26 14:28:31 +0100 (Thu, 26 Feb 2009) | 3 lines

minor commit to test post-commit script changes


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r178704 | kpfleming | 2009-02-26 14:39:00 +0100 (Thu, 26 Feb 2009) | 4 lines

another minor commit to test post-commit script changes (now testing post-revprop-change as well, third try)



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r178733 | oej | 2009-02-26 16:02:53 +0100 (Thu, 26 Feb 2009) | 2 lines

Clarifications on the different models and reference to further docs.

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r178764 | file | 2009-02-26 16:40:10 +0100 (Thu, 26 Feb 2009) | 5 lines

Ensure there is a valid tone part before trying to play tones.

(closes issue #14558)
Reported by: alecdavis

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r178767 | dvossel | 2009-02-26 16:50:22 +0100 (Thu, 26 Feb 2009) | 8 lines

IAX2 prune realtime fix

Iax2 prune realtime had issues.  If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened.  This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing.  If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine.  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.  

(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/

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r178801 | file | 2009-02-26 17:42:36 +0100 (Thu, 26 Feb 2009) | 5 lines

Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed.

(closes issue #14541)
Reported by: grant

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r178828 | murf | 2009-02-26 18:22:11 +0100 (Thu, 26 Feb 2009) | 34 lines

Merged revisions 178804 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
  
  This patch prevents the feature detection timeout from being cut in half.
  
  Because the ast_channel_bridge() call will return 0 and pass
  a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
  field in hte config struct is getting decremented twice, which 
  effectively cuts the digittimeout in half. I added conditions
  to the if statement to only let DTMF_END frames to flow thru,
  which solved the problem. Also, when the frame pointer is null,
  let control flow thru-- this usually happens on timeouts. I added
  a comment to the code to explain what's going on and why.
  
  Many thanks to sodom for reporting this problem. Personnally, it always seemed
  like something was wrong with the featuredigittimeout, but I never
  could quite decide what... and was too busy to investigate.
  This bug forced the issue, and now we know.
  
  Sodom had other issues in 14515, but I couldn't reproduce them. If
  he still has problems, and wants to get them solved, he is welcome
  to reopen 14515.
  
  
  (closes issue #14515)
  Reported by: sodom
  Patches:
        14515.patch uploaded by murf (license 17)
  Tested by: murf, sodom
........

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r178848 | dvossel | 2009-02-26 18:26:04 +0100 (Thu, 26 Feb 2009) | 14 lines

Blocked revisions 178838 via svnmerge

........
  r178838 | dvossel | 2009-02-26 11:24:02 -0600 (Thu, 26 Feb 2009) | 9 lines
  
  IAX2 prune realtime fix
  
  Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime.  These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend.  For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
  
  (closes issue #14479)
  Reported by: mousepad99
  Review: http://reviewboard.digium.com/r/176/
........

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r178870 | murf | 2009-02-26 18:45:22 +0100 (Thu, 26 Feb 2009) | 1 line

These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x.
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r178871 | dvossel | 2009-02-26 18:46:12 +0100 (Thu, 26 Feb 2009) | 6 lines

IAX2 prune realtime, minor tweak to last fix

A return statement was missing which caused unexpected cli output.

issue #14479

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r178919 | tilghman | 2009-02-26 19:41:28 +0100 (Thu, 26 Feb 2009) | 8 lines

Sound confirmation of call pickup success.
(closes issue #13826)
 Reported by: azielke
 Patches: 
       pickupsound2-trunk.patch uploaded by azielke (license 548)
       __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10)
 Tested by: lmadsen

................
r178986 | murf | 2009-02-27 04:45:58 +0100 (Fri, 27 Feb 2009) | 26 lines

Merged revisions 178956 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.

........
  r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
  
  This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
  
  As per bug 14515, a dev discussion arrived at a "mediated concensus" 
  of a default feature digit timeout of 1.0 sec. Some voted for 1300;
  ctooley thought 1500 for distracted phone users in phone booths; 
  kpfleming put his foot down at 1.0 sec. 
  
  Users who found the previous default max delay of 250 msec perfect,
  are welcome to override the new default. Notice that I said that
  250 msec was the default; wait a minute, you might say, the config
  file said it was 500 msec!; well, because of the bug fix for 14515,
  we found that 500 msec was actually enforcing a max of 250. The bug
  fix would restore 500 msec, but we felt even that was a bit tight
  for most users... 2000 msec was pushed earlier by mmichelson, so
  that reduces to 1000 msec after the bug fix. Enjoy!
........

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r179021 | russell | 2009-02-27 16:51:56 +0100 (Fri, 27 Feb 2009) | 7 lines

Fix downloading SIREN7 and SIREN14 sound packages.

In passing, also fix downloading SLIN16 extra sound packages.

(closes issue #14565)
Reported by: jtodd

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r179057 | qwell | 2009-02-27 20:04:57 +0100 (Fri, 27 Feb 2009) | 8 lines

Update documentation for DIALEDTIME and ANSWEREDTIME variables.

(closes issue #14566)
Reported by: klaus3000
Patches:
      ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65)
      ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65)

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r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines

Add reload support to chan_skinny.

Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.

(closes issue #10297)
Reported by: DEA
Patches:
      skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
      skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
	  With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj

Review: http://reviewboard.digium.com/r/130/

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r179154 | russell | 2009-02-27 22:23:12 +0100 (Fri, 27 Feb 2009) | 2 lines

Add a note about the ordering of entries in sip.conf in 1.6.1.

................
r179161 | tilghman | 2009-02-27 22:32:13 +0100 (Fri, 27 Feb 2009) | 3 lines

If config file is blank, don't load module.
(Closes issue #14563)

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r179164 | russell | 2009-02-27 22:47:18 +0100 (Fri, 27 Feb 2009) | 2 lines

Mark res_ais as experimental, as the binary event format is subject to change.

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r179219 | mmichelson | 2009-03-01 22:45:08 +0100 (Sun, 01 Mar 2009) | 18 lines

Properly free memory and remove scheduler entries when a transmission failure occurs.

Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit 
was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.

XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.

(closes issue #14455)
Reported by: Nick_Lewis
Patches:
      14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis


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r179254 | mmichelson | 2009-03-02 00:25:23 +0100 (Mon, 02 Mar 2009) | 5 lines

Swap reversed timevals.

This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!


................
r179291 | file | 2009-03-02 15:13:45 +0100 (Mon, 02 Mar 2009) | 7 lines

Fix issue where changing the volume of both directions of audio did not work.

(closes issue #14574)
Reported by: KNK
Patches:
      audiohook_volume_fix.diff uploaded by KNK (license 545)

................
r179323 | file | 2009-03-02 15:28:09 +0100 (Mon, 02 Mar 2009) | 5 lines

Do not try to remove a registration scheduled item if the scheduler context has already been destroyed.

(closes issue #14580)
Reported by: alecdavis

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r179361 | tilghman | 2009-03-02 18:18:48 +0100 (Mon, 02 Mar 2009) | 2 lines

Backport 1.6.0 fix to trunk (failsafe if db is not loaded)

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r179396 | qwell | 2009-03-02 21:16:51 +0100 (Mon, 02 Mar 2009) | 12 lines

Merged revisions 179395 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line
  
  Remove several silly warnings in editline.  One about a broken preprocessor directive, and another about strlcpy/strlcat.

  (closes issue #14264)
  Reported by: dimas
........

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r179462 | russell | 2009-03-03 00:00:30 +0100 (Tue, 03 Mar 2009) | 16 lines

Merged revisions 179461 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines

Ensure that only one thread is calling ast_settimeout() on a channel at a time.

For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.

(Found in a debugging session with dvossel and mmichelson)

........

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r179465 | russell | 2009-03-03 00:06:16 +0100 (Tue, 03 Mar 2009) | 4 lines

Fix a reference leak in timerfd_set_rate().

(found during a debugging session with dvossel and mmichelson.)

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r179469 | tilghman | 2009-03-03 00:10:18 +0100 (Tue, 03 Mar 2009) | 17 lines

Merged revisions 179468 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
  
  When ending a recording with silence detection, remember to reduce the duration.
  The end of the recording is correspondingly trimmed, but the duration was not
  trimmed by the number of seconds trimmed, so the saved duration was necessarily
  longer than the actual soundfile duration.
  (closes issue #14406)
   Reported by: sasargen
   Patches: 
         20090226__bug14406.diff.txt uploaded by tilghman (license 14)
   Tested by: sasargen
........

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r179533 | russell | 2009-03-03 00:36:38 +0100 (Tue, 03 Mar 2009) | 48 lines

Merged revisions 179532 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines

Move ast_waitfor() down to avoid the results of the API call becoming stale.

This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice.  By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.

So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available.  Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.

This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk.  He was using the timerfd timing module.  When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was 
the cause of the last legitimate call to ast_read() done by autoservice.  

In this test, an IAX2 channel was calling into the MeetMe conference.  It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled.  Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled.  So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.

Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed.  When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function.  The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read.  This caused Asterisk
to lock up very quickly.

Thanks to dvossel and mmichelson for the fun debugging session.  :-)

........

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r179537 | jpeeler | 2009-03-03 01:01:51 +0100 (Tue, 03 Mar 2009) | 16 lines

Merged revisions 179536 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines
  
  Fix bridging regression from commit 176701
  
  This fixes a bad regression where the bridge would exit after an attended
  transfer was made. The problem was due to nexteventts getting set after the
  masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
  
  (closes issue #14315)
  Reported by: tim_ringenbach
........

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r179609 | russell | 2009-03-03 14:54:41 +0100 (Tue, 03 Mar 2009) | 17 lines

Merged revisions 179608 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines

Make it easier to detect an improper call to ast_read().

When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno.  This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.