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Author SHA1 Message Date
tilghman dbcca86ee1 An offhand comment from Russell made me realize that the configuration file
caching would not work properly for users.conf and any other file read from
more than one place.  I needed to add the filename which requested the config
file to get it to work properly.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107791 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 22:55:16 +00:00
file 0d41e1f3e5 Merged revisions 107646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 lines

Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted.
(closes issue #11707)
Reported by: iam

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107659 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 19:23:28 +00:00
file 4b2180b9b1 Clarify comment about masquerading and playback of the parking slot.
(closes issue #12180)
Reported by: davidw


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107465 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 15:05:17 +00:00
kpfleming c4402dfe48 Merged revisions 107408 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines

check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)

(closes issue #12179)
Reported by: Netview

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107409 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 14:09:49 +00:00
kpfleming 00544b6a26 Merged revisions 107352 via svnmerge from
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107373 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 11:36:51 +00:00
tilghman d7a3bcf49b (closes issue #6019)
Reported by: ssokol
 Patches: 
       20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
 Tested by: putnopvut


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107231 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 21:48:20 +00:00
russell e5a1a45234 Merged revisions 107161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107162 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 20:17:37 +00:00
russell 8948f4573a Merged revisions 107158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107159 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 20:05:12 +00:00
russell 5f981f0c5f Merged revisions 107102 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107103 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 17:13:34 +00:00
russell 961c14b7a6 Merged revisions 107099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107100 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 16:59:13 +00:00
murf f96fee046b 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107019 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 14:55:21 +00:00
file a4ae0883c2 Merged revisions 107016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107017 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10 14:36:16 +00:00
qwell 3f49dc54f0 Merged revisions 106842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106843 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 22:15:20 +00:00
file 9bde06d5a9 Merged revisions 106788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106789 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 19:33:09 +00:00
murf 2be361fbb9 (closes issue #6002)
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?




git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106757 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 18:57:57 +00:00
tilghman d46f0daab8 Merged revisions 106606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106607 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 15:22:34 +00:00
tilghman 84aa522629 Merged revisions 106552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106553 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 06:54:47 +00:00
russell 5ffedddec1 Merge changes from team/russell/g722-sillyness ...
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: http://svn.digium.com/svn/asterisk/trunk@106501 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 00:24:58 +00:00
qwell e278780384 Fix file playback in many cases.
(closes issue #12115)
Reported by: pj
Patches:
      v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me)
Tested by: dimas, qwell, russell


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106439 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-06 22:11:30 +00:00
mmichelson d3dcb497cd Merged revisions 106437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106438 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-06 22:11:26 +00:00
file f6b76699b7 Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106239 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 22:43:22 +00:00
mmichelson f523ddafbb Adding the Atxfer manager command. With this, you may initiate
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



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106236 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 22:33:05 +00:00
file 2e5953ed87 Fix code up to what it was meant to be.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106110 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 16:39:22 +00:00
tilghman 198829f2db Create a centralized configuration option for silencethreshold
(closes issue #11236)
 Reported by: philipps
 Patches: 
       20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
 Tested by: philipps


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106072 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 16:23:44 +00:00
russell 6d9959cc34 Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines

Fix a bug that I just noticed in the RTP code.  The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105933 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 01:54:16 +00:00
mmichelson 8e4b2b7d21 There are several places in manager.c where BUFSIZ is used for a buffer
which will contain nowhere near that amount of data. This makes these buffers
more reasonably sized.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105864 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 23:24:56 +00:00
tilghman 701a8a40c2 Fix minor misuses of snprintf
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 23:10:45 +00:00
tilghman 832983e43a Whitespace changes only
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105840 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 23:04:29 +00:00
russell de432dde0b add a destroy API call for a server instance
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105804 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 22:28:03 +00:00
russell c2841b668a More public API name changes to use an appropriate ast_ prefix
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105785 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 22:23:21 +00:00
russell 8ef91aad9e Rename public object server_instance to ast_tcptls_server_instance
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105773 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 22:15:18 +00:00
file 320e8ef1d5 Merged revisions 105676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines

In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105677 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 18:11:38 +00:00
file 151876ff4c Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines

When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105675 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 18:08:42 +00:00
russell c58ef902bc Make it so you don't have to cast away const in a couple places
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105594 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 04:47:32 +00:00
russell 9e92942e58 remove unnecessary casts
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105593 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 04:44:28 +00:00
russell d8799569cb - Add curly braces around the while loop
- Properly break out of the loop on error when an included context is not found


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105590 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 04:28:48 +00:00
russell e1a4a45eef Use ast_copy_string() instead of strncpy(), and use sizeof() instead of
a magic number


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105589 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04 04:26:39 +00:00
russell 193c03465d Merged revisions 105565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008) | 3 lines

Update the copyright information for autoservice.  Most of the code in this file
now is stuff that I have written recently ...

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105566 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 16:02:19 +00:00
russell 9b1db53c3e 3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST.  The way this list works makes it such that
   a RWLIST provides no additional benefit.  Also, a mutex is needed for
   use with the thread condition.


Merged revisions 105563 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines

Merge in some changes from team/russell/autoservice-nochans-1.4

These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.

1) Change it so that autoservice thread doesn't keep looping around calling
   ast_waitfor_n() on 0 channels twice a second.  Instead, use a thread condition
   so that the thread properly goes to sleep and does not wake up until a
   channel is put into autoservice.

   This actually fixes an interesting bug, as well.  If the autoservice thread
   is already running (almost always is the case), then when the thread goes
   from having 0 channels to have 1 channel to autoservice, that channel would
   have to wait for up to 1/2 of a second to have the first frame read from it.

2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
   channels and no fds to poll() on, such as was the case with the previous code
   for the autoservice thread.  In this case, the code would call alloca(0), and
   pass the result as the first argument to poll().  In this case, the 2nd
   argument to poll() specified that there were no fds, so this invalid pointer
   shouldn't actually get dereferenced, but, this code makes it explicit and
   ensures the pointers are NULL unless we have valid data to put there.

(related to issue #12116)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 15:59:50 +00:00
file ae272b289b Merged revisions 105560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 lines

It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL.
(closes issue #11911)
Reported by: edgreenberg
Patches:
      v1-11911.patch uploaded by dimas (license 88)
Tested by: tbsky

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105561 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03 15:30:29 +00:00
file 9175fcb897 Add support for 16KHz signed linear.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105509 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-01 03:59:41 +00:00
twilson 50fe8fd5ec Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue.
(closes issue #11520)
Reported by: pliew
Tested by: otherwiseguy


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-01 01:30:37 +00:00
russell c8f1d489b4 Merged revisions 105409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) | 23 lines

Fix a major bug in autoservice.  There was a race condition in the handling of
the list of channels in autoservice.  The problem was that it was possible for
a channel to get removed from autoservice and destroyed, while the autoservice
thread was still messing with the channel.  This led to memory corruption, and
caused crashes.  This explains multiple backtraces I have seen that have
references to autoservice, but do to the nature of the issue (memory corruption),
could cause crashes in a number of areas.

(fixes the crash in BE-386)
(closes issue #11694)
(closes issue #11940)

The following issues could be related.  If you are the reporter of one of these,
please update to include this fix and try again.

(potentially fixes issue #11189)
(potentially fixes issue #12107)
(potentially fixes issue #11573)
(potentially fixes issue #12008)
(potentially fixes issue #11189)
(potentially fixes issue #11993)
(potentially fixes issue #11791)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105410 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-29 23:36:46 +00:00
russell 020573b8ac Merged revisions 105116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines

Fix a bug in the lock tracking code that was discovered by mmichelson.  The issue
is that if the lock history array was full, then the functions to mark a lock as
acquired or not would adjust the stats for whatever lock is at the end of the array,
which may not be itself.  So, do a sanity check to make sure that we're updating
lock info for the proper lock.

(This explains the bizarre stats on lock #63 in BE-396, thanks Mark!)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105144 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-28 22:39:26 +00:00
mmichelson 6e02d54e7b Merged revisions 104841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines

Two fixes:

1. Make the list of ast_dial_channels a lockable list. This is because in some cases,
   the ast_dial may exist in multiple threads due to asynchronous execution of its application, and
   I found some cases where race conditions could exist.

2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since
   it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been
   cleared yet.

(closes issue #12038)
Reported by: jvandal
Patches:
      12038v2.patch uploaded by putnopvut (license 60)
Tested by: jvandal


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105060 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-28 20:14:04 +00:00
qwell e1ed25d450 Merged revisions 105005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines

Make pbx_exec pass an empty string into applications, if we get NULL.
This protects against possible segfaults in applications that may try
 to use data before checking length (ast_strdupa'ing it, for example)

(closes issue #12100)
Reported by: foxfire
Patches:
      12100-nullappargs.diff uploaded by qwell (license 4)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105006 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-28 19:21:15 +00:00
tilghman 85434616d9 Merged revisions 104868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines

Compatibility fix for PPC64
(closes issue #12081)
 Reported by: jcollie
 Patches: 
       asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
 Tested by: jcollie, Corydon76

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104869 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-28 00:11:31 +00:00
mmichelson 2a017cb36c Merged revisions 104783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb 2008) | 4 lines

Bump a couple of more buffers up by 2 so that annoying warnings aren't generated
like crazy on every fileexists_core call.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104784 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 20:37:32 +00:00
tilghman 8b6f404342 Merged revisions 104704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008) | 2 lines

Ensure the session ID can't be 0.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104705 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 18:20:35 +00:00
file de520fcf45 Merged revisions 104665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 lines

Bump up the buffer by 2.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104687 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 17:45:55 +00:00