No subject

........

................
r179672 | file | 2009-03-03 15:40:04 +0100 (Tue, 03 Mar 2009) | 10 lines

Merged revisions 179671 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines
  
  Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
  We have to do this as the underlying channel driver may need the fdno value to determine what to read.
........

................
r179675 | oej | 2009-03-03 16:13:42 +0100 (Tue, 03 Mar 2009) | 2 lines

Please prefix default values with DEFAULT

................
r179742 | russell | 2009-03-03 17:47:28 +0100 (Tue, 03 Mar 2009) | 14 lines

Merged revisions 179741 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines

Ensure chan->fdno always gets reset to -1 after handling a channel fd event.

Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to.  So, set it to -1 in a few other places, too.

........

................
r179745 | mmichelson | 2009-03-03 18:03:47 +0100 (Tue, 03 Mar 2009) | 8 lines

Convert pbx_spool to use string fields instead of statically-sized buffers.

In tests run after making this conversion, I noticed an approximate 85% 
reduction in memory usage for call file processing.

Review: http://reviewboard.digium.com/r/168/


................
r179841 | file | 2009-03-03 19:28:46 +0100 (Tue, 03 Mar 2009) | 16 lines

Merged revisions 179840 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines
  
  Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
  
  It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
  the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
  We can not safely modify it afterwards because of this, so don't even try.
  
  (closes issue #14564)
  Reported by: meric
........

................
r179903 | bmd | 2009-03-03 21:02:20 +0100 (Tue, 03 Mar 2009) | 1 line

fix a leaked channel lock (and future deadlock) when we try to pick up our own channel
................
r179937 | mmichelson | 2009-03-03 21:59:16 +0100 (Tue, 03 Mar 2009) | 20 lines

Add documentation for timing modules used in Asterisk

This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.

I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.

Big thanks to all who contributed comments on this.

(closes issue #14490)
Reported by: mmichelson
Patches:
      timing.txt uploaded by mmichelson (license 60)

Review: http://reviewboard.digium.com/r/164/

................
r179972 | dvossel | 2009-03-03 23:01:24 +0100 (Tue, 03 Mar 2009) | 13 lines

app_meetme not setting filename and fileformat correctly for realtime

When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set.  Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. 

(closes issue #14545)
Reported by: dalbaech
Patches:
	app_meetme-realtime5.patch uploaded by dvossel (license 671)
	Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705)
Tested by: dvossel, dalbaech
Review: http://reviewboard.digium.com/r/180/


................
r179973 | murf | 2009-03-03 23:12:02 +0100 (Tue, 03 Mar 2009) | 33 lines

Merged revisions 179807 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

I had some work to do to port these changes to trunk; the 
check_expr stuff hasn't been updated here for quite some
time, it appears. I added some more tests to the check_expr2
suite. I had to play around with the makefile a bit, etc.

I added STANDALONE2 #ifdefs to ast_expr2.y so as not to
conflict structure with aelparse.

........
  r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines
  
  These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
  
  I modified and added rules in ast_expr2.fl to better handle
  the concatenations.
  
  I added some default routines to ast_expr2.y so the standalone would
  compile. It also looks like I haven't run this thru bison since 2.1, so
  it's good to get this updated.
  
  The Makefile has comments added now for check_expr2 and check_expr to
  explain what they are for, and how to run them. 
  
  The testexpr2s stuff has been removed, in favor of check_expr2.
  
  expr2.testinput has been updated to include the two expressions
  that inspired these changes (from mcnobody on #asterisk this morning)
  The regression has been run and all looks well.
........

................
r180007 | mmichelson | 2009-03-03 23:49:07 +0100 (Tue, 03 Mar 2009) | 22 lines

Merged revisions 180006 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
  
  Clarify some documentation of queues.conf.sample
  
  It had always been possible to explicitly specify a "blank"
  value for a sound file in queues.conf and have no sound played
  back. The problem with this is that it would result in some ugly
  CLI warnings from file.c.
  
  This commit introduces a check when playing a file in app_queue

[... 62210 lines stripped ...]