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

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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines

fix up various compiler warnings found with gcc-4.3:

- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)

- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement

- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur

- main/editline/readline.c had an unused variable


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r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines

If we fail to alloc a channel, we should re-lock the pvt structure before returning.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 Reported by: ssokol
 Patches: 
       20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
 Tested by: putnopvut


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r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines

Make sure to reenable echo can after a "failed" (canceled, etc) three-way call.

(closes issue #11335)
Reported by: rebuild

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r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines

Fix another bug specifically related to asynchronous call origination.  Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread.  We can no longer access it in this
code.  If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.

(inspired by BE-386)

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r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines

Fix some bugs related to originating calls.  If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed.  This caused weird memory corruption and
crashes to occur.

(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)

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If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677)
Reported by: alex-911


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r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) | 2 lines

Resolve a compiler warning.

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r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) | 3 lines

Fix a race condition where the generator can go away
(closes issue #12175, reported by edantie, patched by me)

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was the way that locks are referenced, since the old 1.2 names were still used
in the comments.

(closes issue #11997)
Reported by: snuffy
Patches:
      bug_11997_queue_doxy.diff uploaded by snuffy (license 35)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107068 65c4cc65-6c06-0410-ace0-fbb531ad65f3

way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug.

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r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines

Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
(closes issue #11516)
Reported by: ys
Patches:
      branch_1.4_cdr.diff uploaded by ys (license 281)
Tested by: anest, jcapp, dartvader

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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines

don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"

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r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines

Only start the SLA thread if SLA has actually been configured.

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r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines

Fix hardcoded grep in editline, were GNU grep is required.

(closes issue #12124)
Reported by: dmartin

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r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 lines

Ignore source update control frame.
(closes issue #12168)
Reported by: plack

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Reported by: rizzo
Tested by: murf

Proposal of the changes to be made, and then an announcement of how they were accomplished:

http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html

and:

http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html

Here is a recap, file by file, of what I have done:

pbx/pbx_config.c
pbx/pbx_ael.c

All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.

We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.

pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and 
then call merge_contexts_and_delete, which will merge (now) existing contexts and 
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then 
destroy the old dialplan.



chan_sip.c
chan_iax.c
chan_skinny.c

All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.

chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.


apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c

All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.


include/asterisk/pbx.h

ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create()  interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.


include/asterisk/pval.h

ast_compile_ael2() interface changed to include the local hashtab table ptr.


main/features.c

For the sake of the parking context, we use ast_context_find_or_create().



main/pbx.c

I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.

refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.

Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.

Added some calls to ast_verb(3,...) for debug messages

Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.

find_or_create was upgraded to handle both local lists/tables as well as the globals.

context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables

ast_merge_contexts_and_delete() was heavily modified.

ast_add_extension2() was also upgraded to handle changes. 

the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.



res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile

Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps.  The main gotcha was I had to 
include lock.h and hashtab.h in several places.


As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.

How's this for verbose commit messages?




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r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines

Change a warning message to a debug message.  This is happening quite frequently,
and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen.  As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.

(closes issue #12154)

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r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008) | 3 lines

Warn the user when a temporary greeting exists
(Closes issue #11409)

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r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines

Properly initialize rtp->schedid
(Closes issue #12154)

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r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines

Safely use the strncat() function.
(closes issue #11958)
 Reported by: norman
 Patches: 
       20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)

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Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.

main/rtp.c:
 - When a G722 frame is read from the smoother, the number of samples in the
   frame must be divided by 2 before being sent out over the network.  Even
   though G722 is 16 kHz, an error in some previous spec has made it so that
   we have to list the number of samples such as if it was 8 kHz.

main/file.c:
 - When scheduling the next time to expect a frame, take into account that the
   format of the file we're reading from may not be 8 kHz.

codecs/codec_g722.c:
 - When converting from G722 to slinear, g722_decode() expects its samples
   parameter to be in the silly (real samples / 2) format.  Make it so.
 - When converting from slinear to G722, properly set the number of samples in
   the frame to be the number of bytes of output * 2.

formats/format_pcm.c:
 - This format module handles G722, among a number of other formats.  However,
   the read() and seek() functions did not account for the fact that G722 has
   2 samples per byte.

(closes issue #12130, reported by rickross, patched by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12115)
Reported by: pj
Patches:
      v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me)
Tested by: dimas, qwell, russell


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r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines

Quell an annoying message that is likely to print every single time that 
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.

Thanks to joetester on IRC for pointing this out


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trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense 
given our removal of deadagi as an actual application.

(closes issue #12161)
Reported by: explidous
Patches:
      res_agi_12161.patch uploaded by juggie (license 24)
Tested by: juggie


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106399 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12112)
 Reported by: cyrenity
 Patches: 
       res_config_ldap.patch-03-03-2008 uploaded by cyrenity (license 416)
 Tested by: cyrenity, Corydon76


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r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) | 2 lines

Upgrade to the next release of sounds

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was written by seanbright.  It is much sexier than my curses one.  :)

(issue #12139)


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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines

Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas

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r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) | 3 lines

Fix a potential deadlock and a few different potential crashes.
(closes issue #12145, reported by thiagarcia, patched by me)

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an attended transfer over AMI

(closes issue #10585)
Reported by: ornati
Patches:
      atxfer-trunk-r90428.diff uploaded by ornati (license 210)
	  (with modifications from me)
Tested by: putnopvut



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(closes issue #12151)
Reported by: caio1982
Patches:
      DB_metric3.diff uploaded by caio1982 (license 22)


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r106178 | mvanbaak | 2008-03-05 22:12:36 +0100 (Wed, 05 Mar 2008) | 5 lines

document var_metric so no bugreports will come in when it's actually a configuration issue.
(issue #12151)
Reported and patched by: caio1982
1.4 patch by me

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(Closes issue #12147)


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