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

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  r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line
  
  dist-clean should remove dependency information files as well
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  r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line
  
  when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice
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  r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines
  
  major update to doxygen configuration file:
  
  1) update to doxygen 1.5.x style file, as used in trunk
  
  2) tell doxygen where are header files are, so include-file processing can be done
  
  3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function
  
  4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields
  
  5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit)
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the comparison is case insensitive.


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function is case insensitive.


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removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related



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Use Mark's new ast_str_case_hash function instead of jumping through hoops to do insensitive case lookups


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156916 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13903)
 Reported by: kebl0155


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.

This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to 
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.

This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.



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

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r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines

If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.

This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!


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  r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
  
  ast_waitfordigit() requires that the channel be up, for no good logical
  reason.  This prevents While/EndWhile from working within the "h"
  extension.
  Reported by: jgalarneau (for ABE C.2)
  Fixed by: me
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  r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
  
  Provide more space for all the data which can appear in an originating
  channel name.
  (closes issue #13398)
   Reported by: bamby
   Patches: 
         manager.c.diff uploaded by bamby (license 430)
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Reported by: smurfix

This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null.



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'pri set debug' was changed from 'pri debug'.


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not necessary, so I am reverting it



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  - MeetMe()
  - MeetMeCount()
  - MeetMeChannelAdmin()
  - MeetMeAdmin()
  - SLAStation()
  - SLATrunk()

- Add an attribute to optionlist 'hasparams' with the same functionality as the one
we have in <parameter> and <argument> (the DTD was updated)
- Fix a leak when getting an attribute while parsing an <optionlist>.


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we need to pass the name of the node that we are looking for.


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using ':%s/\s\+$//' pointed by seanbright on #asterisk-dev


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

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  r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines
  
  When using call limits under 1 second, infinite call lengths are allowed,
  instead.
  (closes issue #13851)
   Reported by: ruddy
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- Register the aliased cli command (or we will not alias anything).
- Use ARRAY_LEN() when possible.


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r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines

It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev;  he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been. 

This change, tested on these 3 patterns now 
picks the proper one.

However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half 
years or so now.



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  r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines
  
  If the SLA thread is not started, then reload causes a memory leak.
  (closes issue #13889)
   Reported by: eliel
   Patches: 
         app_meetme.c.patch uploaded by eliel (license 64)
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r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines

For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. 


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  r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines
  
  Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not
  to be sent, and instead, schedule a task to destroy the iax2 pvt structure
  10 seconds later.  This allows the IAX2 HANGUP packet to be queued,
  transmitted, and ACKed before the pvt is destroyed.
  (closes issue #13645)
   Reported by: dzajro
   Patches: 
         20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
   Tested by: vazir
   Reviewed: http://reviewboard.digium.com/r/51/
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r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines

(closes issue #13173)
Reported by: pep

This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference.

Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/

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r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines

When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.


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r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines

Move the sanity check that makes sure "always fork" is not set along with the 
console option to be after the code that reads options from asterisk.conf.  
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
 figure out what was causing the 100% CPU problem.)

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we need to free it with ast_free().


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macro. These missing calls were discovered when working on
timerfd support in a separate branch.



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   - database deltree
   - database get
   - exec
   - get data
   - get full variable



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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code

Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.

Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.

ok russellb@ via reviewboard

(closes issue #13735)
Reported by: mvanbaak


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- Add to the DTD the possibility to put a parameter inside an <enum>.


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A new <agi> element is used to describe the XML documentation.
We have the usual synopsis,syntax,description and seealso for AGI commands.
The CLI 'agi show commands' command was changed to show all the documentation se
ctions.



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 changing comment style to conform coding guidelines



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 Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig




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