a comma.  This was causing problems with switch statements with empty expressions.

(closes issue #13901)
Reported by: smurfix
Patches:
      20081118_bug13901.diff uploaded by seanbright (license 71)
Tested by: seanbright
Reviewed by: murf


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https://origsvn.digium.com/svn/asterisk/trunk

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r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines

Begin on a crusade to end trailing whitespace!

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines

We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a 
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.


(closes issue #12994)
Reported by: pabelanger
Patches:
      12994.patch uploaded by putnopvut (license 60)

Closes AST-129


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue #13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage


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Reported by: snyfer

This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play.



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  r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
  
  the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
  
  with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
  
  while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
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'queue remove member'



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res_timing_timerfd



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as an unsigned long long on a 64-bit machine.

Reported by kpfleming on IRC



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querying realtime from the dialplan a little more consistent and easy to use.
The original REALTIME function is preserved, for those who are already
accustomed to that interface.
(closes issue #13651)
 Reported by: Corydon76
 Patches: 
       20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: blitzrage, Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This provides a new timing interface. In order to use it,
you must be running a Linux with a kernel version of
2.6.25 or newer and glibc 2.8 or newer.

This timing interface is a good alternative if a timing
source is necessary (e.g. for IAX trunking) but DAHDI is
otherwise unnecessary for the system.

For now, this commit contains the actual work done in the
res_timing_timerfd branch. There are no notices in the README
or CHANGES files yet, but they will be added in my next commit.

The timing API of Asterisk also needs to have a bit of work done
with regards to choosing which timing interface to use. This commit
makes the choice a build-time decision, by only allowing one of
the timer interfaces to be chosen in menuselect. It would be preferable
if the choice could be made at run-time, however. The preferred timing
interface could be loaded and tested, and if it does not work, choice
number two may be used instead. That sort of thing. That is beyond
the scope of work in this branch though.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157820 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix checking for CONFIG_STATUS_FILEINVALID so that modules don't crash upon trying to parse an invalid config


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(closes issue #13473)
 Reported by: genie
 Patches: 
       20080916__bug13473.diff.txt uploaded by Corydon76 (license 14)


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make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines

also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases



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in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also
deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions.  va_copy() is C99, anyway, and we already require C99 for
other purposes, so this isn't really a big change anyway.  This change also
simplifies some of the core ast_str_* functions.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

else Asterisk will crash when attempting to dereference the NULL
pointer

(closes issue #13858)
Reported by: eliel
Patches:
      astmm.c.patch uploaded by eliel (license 64)



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configure instead of the icky loop I was doing before).

(closes issue #13842)
Reported by: snuffy
Patches:
      bug13842_20081106.diff uploaded by snuffy (license 35)
      13842.diff uploaded by seanbright (license 71)
Tested by: snuffy


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is unloaded and then Asterisk is stopped. The problem was that
we are not unregistering the ast_moh_destroy function at exit.

(closes issue #13761)
Reported by: eliel
Patches:
      res_musiconhold.c.patch uploaded by eliel (license 64)



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target of the Makefile.

(closes issue #13875)
Reported by: pabelanger
Patches:
      Makefile.155928 uploaded by pabelanger (license 224)



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is in use so that the message is deleted from both
local and IMAP storage.

(closes issue #13642)
Reported by: jaroth
Patches:
      deleteyes.patch uploaded by jaroth (license 50)




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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines

Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.

(closes issue #13878)
Reported by: nahuelgreco
Patches:
      sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
	  Tested by: putnopvut

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changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.



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(closes issue #13667)
Reported by: sherif
Patches:
      archlinux_rc_makefile.patch uploaded by sherif (license 591)
      archlinux_rc_makefile-2.patch uploaded by mvanbaak (license 7)


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* Revert logic to mirror 1.4's in the sense that it will not allow
  the call counter to dip below 0.

These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.

(closes issue #13668)
Reported by: mjc
Patches:
      hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines

(closes issue #13899)
Reported by: akkornel

This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines

Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.

This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.


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Reported by: alex70
Patches:
      13420.13539.patch uploaded by murf (license 17)
Tested by: murf, awk

This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only
generated when category had no previous comments.

The other problem: Insertions could get the line-numbering
out of whack and generate negative line numbers, causing 
chunks of line numbers to be emitted, on the scale of the
number of lines up to that point in the file. In such cases,
abort the looping, and all is well.




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a function: in these cases, we have to use the heap, or garbage will result.
(closes issue #13898)
 Reported by: alecdavis
 Patches: 
       20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: alecdavis


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