[asterisk-commits] murf: branch group/CDRfix5 r77812 - in /team/group/CDRfix5: ./ apps/ build_to...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jul 30 23:25:23 CDT 2007
Author: murf
Date: Mon Jul 30 23:25:21 2007
New Revision: 77812
URL: http://svn.digium.com/view/asterisk?view=rev&rev=77812
Log:
Merged revisions 76619,76621,76655,76657,76703-76704,76706-76707,76709-76712,76755,76770,76785,76791,76804,76807,76841,76852,76874,76892,76925,76940,76984-76985,77023,77054,77072,77155-77156,77182,77217-77218,77233,77248,77266-77269,77284,77299,77319,77349,77351,77381,77411,77432,77461,77491,77520,77534,77537-77538,77541,77572,77602-77603,77616,77630-77631,77647-77648,77650,77653-77654,77668-77669,77682,77684,77697,77711-77712,77725-77726,77739,77753,77766,77769-77770,77772-77773,77779,77781,77784,77786-77787,77789-77793,77796-77797,77799-77801,77808,77810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r76619 | file | 2007-07-23 11:49:45 -0600 (Mon, 23 Jul 2007) | 10 lines
Merged revisions 76618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2 lines
Allow app_morsecode to build on PPC Linux by putting the value of the digit char in an int.
........
................
r76621 | qwell | 2007-07-23 11:58:46 -0600 (Mon, 23 Jul 2007) | 13 lines
Merged revisions 76620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10276)
........
r76620 | qwell | 2007-07-23 12:57:53 -0500 (Mon, 23 Jul 2007) | 4 lines
Don't try to queue up hold/unhold frames on a non-existent channel.
Issue 10276.
........
................
r76655 | file | 2007-07-23 12:31:06 -0600 (Mon, 23 Jul 2007) | 20 lines
Merged revisions 76654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, 23 Jul 2007) | 12 lines
Merged revisions 76653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines
(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.
........
................
................
r76657 | qwell | 2007-07-23 13:00:19 -0600 (Mon, 23 Jul 2007) | 11 lines
Merged revisions 76656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul 2007) | 3 lines
Fix some incorrect softkey labels in messages.
Don't try to play dialtone in some unimplemented features.
........
................
r76703 | tilghman | 2007-07-23 13:51:41 -0600 (Mon, 23 Jul 2007) | 6 lines
Merge the dialplan_aesthetics branch. Most of this patch simply converts applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
................
r76704 | tilghman | 2007-07-23 14:27:26 -0600 (Mon, 23 Jul 2007) | 3 lines
Missed one conversion to comma delimiter (thanks, Juggie) and add documentation on the
change to the Local channel name.
................
r76706 | file | 2007-07-23 15:42:43 -0600 (Mon, 23 Jul 2007) | 2 lines
Clean up res_crypto module. It now uses an rwlist to keep the keys and it should also be thread safe now.
................
r76707 | tilghman | 2007-07-23 16:02:05 -0600 (Mon, 23 Jul 2007) | 8 lines
Enhance AGI with several fixes:
- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
................
r76709 | tilghman | 2007-07-23 16:41:27 -0600 (Mon, 23 Jul 2007) | 12 lines
Merged revisions 76708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) | 4 lines
It was our stated intention for 1.4 that files created in app_voicemail should
depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600,
regardless of the umask. This corrects that deficiency.
........
................
r76710 | file | 2007-07-23 17:05:18 -0600 (Mon, 23 Jul 2007) | 2 lines
Don't bother calling AST_RWLIST_EMPTY on a list before AST_RWLIST_TRAVERSE, it's just a double check.
................
r76711 | file | 2007-07-23 17:14:20 -0600 (Mon, 23 Jul 2007) | 2 lines
You need to put static in front of a static RWLIST declaration to make it really static... and don't call AST_RWLIST_HEAD_DESTROY on a statically declared list.
................
r76712 | file | 2007-07-23 20:59:49 -0600 (Mon, 23 Jul 2007) | 2 lines
Move manager users list over to an rwlist.
................
r76755 | rizzo | 2007-07-24 01:51:14 -0600 (Tue, 24 Jul 2007) | 3 lines
add documentation on nat/stun support in chan_sip
................
r76770 | rizzo | 2007-07-24 08:49:49 -0600 (Tue, 24 Jul 2007) | 5 lines
two small fixes when using stun (reported by Marta Carbone):
+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload
................
r76785 | qwell | 2007-07-24 09:35:58 -0600 (Tue, 24 Jul 2007) | 10 lines
The chan_skinny Dial() syntax was funky. You had to do Dial(Skinny/line at device)
This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.
Note that this does not remove or deprecate the "old" syntax, as it's still
quite useful - even moreso if shared lines get implemented.
Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)
................
r76791 | file | 2007-07-24 10:09:20 -0600 (Tue, 24 Jul 2007) | 2 lines
Don't download/install the sound packages if already installed.
................
r76804 | mmichelson | 2007-07-24 10:42:36 -0600 (Tue, 24 Jul 2007) | 21 lines
Merged revisions 76801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul 2007) | 13 lines
Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue.
This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is
roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is
used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone.
As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through
the member list to figure out how many members there are.
Special thanks to blitzrage for helping to test this out.
(closes issue #10127, reported by bcnit, patched by me, tested by blitzrage)
........
................
r76807 | tilghman | 2007-07-24 11:05:10 -0600 (Tue, 24 Jul 2007) | 11 lines
Merged revisions 76803 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 lines
Don't create the Asterisk channel until we are starting the PBX on it.
(ASA-2007-018)
........
................
r76841 | qwell | 2007-07-24 11:23:16 -0600 (Tue, 24 Jul 2007) | 2 lines
Trivial whitespace change to test comitting...
................
r76852 | qwell | 2007-07-24 11:43:36 -0600 (Tue, 24 Jul 2007) | 1 line
Revert trivial whitespace change (for testing)
................
r76874 | tilghman | 2007-07-24 12:19:18 -0600 (Tue, 24 Jul 2007) | 2 lines
Fix escaping and some of the formattting (closes issue #10285)
................
r76892 | tilghman | 2007-07-24 14:45:32 -0600 (Tue, 24 Jul 2007) | 9 lines
Blocked revisions 76891 via svnmerge
........
r76891 | tilghman | 2007-07-24 15:42:05 -0500 (Tue, 24 Jul 2007) | 2 lines
Found another place where we should be using the umask (thanks jcmoore)
........
................
r76925 | tilghman | 2007-07-24 15:37:11 -0600 (Tue, 24 Jul 2007) | 2 lines
Add the flag to trigger an intentional crash on mutex errors
................
r76940 | tilghman | 2007-07-24 16:13:37 -0600 (Tue, 24 Jul 2007) | 18 lines
Merged revisions 76937 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r76937 | tilghman | 2007-07-24 17:12:43 -0500 (Tue, 24 Jul 2007) | 10 lines
Merged revisions 76934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 Jul 2007) | 2 lines
Oops, res contains the error code, not errno. I was wondering why a mutex was reporting "No such file or directory"...
........
................
................
r76984 | murf | 2007-07-24 18:34:42 -0600 (Tue, 24 Jul 2007) | 17 lines
Merged revisions 76983 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue, 24 Jul 2007) | 9 lines
Merged revisions 76978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line
this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing
........
................
................
r76985 | russell | 2007-07-24 19:06:02 -0600 (Tue, 24 Jul 2007) | 1 line
remove a couple of entries that got duplicated and snuck into the SIP section. Also, align the NAT/STUN entry with the others.
................
r77023 | rizzo | 2007-07-25 03:45:15 -0600 (Wed, 25 Jul 2007) | 11 lines
Merged revisions 77022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 lines
set the sequence number in a frame for all frame types
........
................
r77054 | rizzo | 2007-07-25 08:13:17 -0600 (Wed, 25 Jul 2007) | 3 lines
change the debug level to 3 for an exceedingly annoying message
(3-deep nested loop)
................
r77072 | file | 2007-07-25 11:16:11 -0600 (Wed, 25 Jul 2007) | 10 lines
Merged revisions 77071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul 2007) | 2 lines
Fix autoconf logic for finding OpenH323 when it is not in the first place searched (/usr/share/openh323).
........
................
r77155 | mmichelson | 2007-07-25 15:53:35 -0600 (Wed, 25 Jul 2007) | 11 lines
Merged revisions 77154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines
chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string
........
................
r77156 | rizzo | 2007-07-25 15:58:13 -0600 (Wed, 25 Jul 2007) | 8 lines
silence a warning in ast-devmode on a potentially uninitialized var.
At first sight (but the function is very large so i am not 100% sure)
the code seems correct, so maybe my compiler is just not smart
enough to figure that out at the optimization level it has.
Not worthwhile merging to 1.4 i believe.
................
r77182 | file | 2007-07-25 16:18:56 -0600 (Wed, 25 Jul 2007) | 12 lines
Merged revisions 77176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines
(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.
........
................
r77217 | murf | 2007-07-25 19:13:07 -0600 (Wed, 25 Jul 2007) | 9 lines
Merged revisions 77191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
........
................
r77218 | murf | 2007-07-25 19:33:55 -0600 (Wed, 25 Jul 2007) | 1 line
The upgrade of application argument separators to comma has an effect on AEL; I commented out the code that substitutes commas with vertbars, so we can get apps to parse their args correctly.
................
r77233 | rizzo | 2007-07-25 22:47:54 -0600 (Wed, 25 Jul 2007) | 2 lines
add an entry for h263plus in an empty slot of the rtp types.
................
r77248 | rizzo | 2007-07-25 23:35:42 -0600 (Wed, 25 Jul 2007) | 4 lines
document how the RTP marker bit is passed for video frames,
and why this does not overwrite useful information.
................
r77266 | russell | 2007-07-26 07:10:49 -0600 (Thu, 26 Jul 2007) | 2 lines
Add a link to the list of assigned RTP payload types for convenience.
................
r77267 | tilghman | 2007-07-26 07:19:07 -0600 (Thu, 26 Jul 2007) | 2 lines
Things expecting a positive result from ast_random() should not be surprised (closes #10308)
................
r77268 | russell | 2007-07-26 07:20:36 -0600 (Thu, 26 Jul 2007) | 3 lines
Ensure that the read from /dev/urandom returns a positive result
(closes issue #10308, reported by yehavi, patched by me)
................
r77269 | russell | 2007-07-26 07:26:44 -0600 (Thu, 26 Jul 2007) | 4 lines
Revert some changes to call abs() on the result of ast_random().
* random() is defined to return a positive result, and now ast_random()
will always do so as well
................
r77284 | russell | 2007-07-26 08:49:51 -0600 (Thu, 26 Jul 2007) | 5 lines
Merge a big batch of documentation fixes for escaping, marking URLs, places
where verbatim text went off the end of the page on the PDF, and various
other improvements
(closes issue #10307, IgorG)
................
r77299 | russell | 2007-07-26 09:49:18 -0600 (Thu, 26 Jul 2007) | 12 lines
Do a massive conversion for using the ast_verb() macro
(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
................
r77319 | mmichelson | 2007-07-26 12:31:28 -0600 (Thu, 26 Jul 2007) | 16 lines
Merged revisions 77318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul 2007) | 8 lines
Two consecutive calls to PQfinish could occur, meaning free gets called on the same variable twice.
This patch sets the connection to NULL after calls to PQfinish so that the problem does not occur.
Also in this patch, prashant_jois informed me that it is safe to pass a null pointer to PQfinish, so
I have removed the check for conn's existence from my_unload_module.
(closes issue 10295, reported by junky, patched by me with input from prashant_jois)
........
................
r77349 | tilghman | 2007-07-26 13:29:12 -0600 (Thu, 26 Jul 2007) | 10 lines
Merged revisions 77348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007) | 2 lines
Oops, that builtin define should be all-lowercase.
........
................
r77351 | tilghman | 2007-07-26 13:33:47 -0600 (Thu, 26 Jul 2007) | 10 lines
Merged revisions 77350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007) | 2 lines
Missed one
........
................
r77381 | mmichelson | 2007-07-26 14:39:46 -0600 (Thu, 26 Jul 2007) | 15 lines
Merged revisions 77380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 Jul 2007) | 7 lines
Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never
attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were
made to acccomodate 64 bit systems in ast_backtrace.
Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed
........
................
r77411 | russell | 2007-07-26 15:24:42 -0600 (Thu, 26 Jul 2007) | 18 lines
Merged revisions 77410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) | 10 lines
AST_DEVMODE was defined in trunk, but not in 1.4. When Asterisk is compiled
under dev mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
define it in the same way that trunk does. Also, revert the change that added
this define in the Makefile
The advantage to doing it this way is that buildopts.h gets installed when
you install Asterisk. Then, when building any out of tree modules, or
building asterisk-addons, these modules know which options the rest of Asterisk
was built with.
........
................
r77432 | kpfleming | 2007-07-26 16:17:25 -0600 (Thu, 26 Jul 2007) | 15 lines
Merged revisions 77424,77429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007) | 2 lines
use new canonical name for download server
........
r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007) | 2 lines
change protocol for downloads as well
........
................
r77461 | file | 2007-07-26 17:20:25 -0600 (Thu, 26 Jul 2007) | 12 lines
Merged revisions 77460 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines
(closes issue #10302)
Reported by: litnialex
If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already.
........
................
r77491 | mmichelson | 2007-07-27 08:31:35 -0600 (Fri, 27 Jul 2007) | 11 lines
Merged revisions 77490 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines
"re-invite" was misspelled
........
................
r77520 | murf | 2007-07-27 09:46:20 -0600 (Fri, 27 Jul 2007) | 1 line
These fixes take care of two problems: a complaint in asterisk-dev that goto's aren't working in trunk, a side effect of the move to commas as arg seps in apps and funcs; and a problem I spotted myself with dial's 'e' option, where gotos were off by one, because I forgot to set the AUTOLOOP flag in the peer channel.
................
r77534 | tilghman | 2007-07-27 10:20:55 -0600 (Fri, 27 Jul 2007) | 2 lines
'dialplan save' shouldn't be converting '|' back to ',' anymore.
................
r77537 | file | 2007-07-27 10:29:40 -0600 (Fri, 27 Jul 2007) | 14 lines
Merged revisions 77536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6 lines
(closes issue #10323)
Reported by: julianjm
Patches:
chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99)
Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing.
........
................
r77538 | file | 2007-07-27 10:31:55 -0600 (Fri, 27 Jul 2007) | 2 lines
Turn 4 lines of code into 1 line that does the same thing.
................
r77541 | file | 2007-07-27 11:05:18 -0600 (Fri, 27 Jul 2007) | 14 lines
Merged revisions 77540 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77540 | file | 2007-07-27 14:04:08 -0300 (Fri, 27 Jul 2007) | 6 lines
(closes issue #10310)
Reported by: prashant_jois
Patches:
cdr_pgsql.patch uploaded by prashant (license 114)
Finish the Postgresql connection after the log messages are printed so we don't access invalid memory.
........
................
r77572 | tilghman | 2007-07-27 12:17:12 -0600 (Fri, 27 Jul 2007) | 10 lines
Merged revisions 77571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77571 | tilghman | 2007-07-27 13:15:58 -0500 (Fri, 27 Jul 2007) | 2 lines
Missing newline
........
................
r77602 | tilghman | 2007-07-27 17:16:04 -0600 (Fri, 27 Jul 2007) | 2 lines
Target asterisk.pdf stopped building when the build was moved to the doc directory.
................
r77603 | tilghman | 2007-07-27 17:21:23 -0600 (Fri, 27 Jul 2007) | 2 lines
Some ODBC drivers don't set the CHAR_OCTET_LENGTH field correctly.
................
r77616 | rizzo | 2007-07-28 01:44:16 -0600 (Sat, 28 Jul 2007) | 27 lines
make use of received= and rport= fields in sip replies.
In a nutshell, these fields are used to tell a sip entity
the address and port its request came from, and are extremely
useful in the presence of NATs, especially with symmetric NATs
where STUN is totally ineffective.
This patch stores the address and port in the 'ourip' field of
the dialog descriptor, so they can be reused in subsequent transactions.
As it is, it works well for things like REGISTER requiring authentication,
because the second REGISTER request (with auth credentials) will carry
the correct address. Maybe it can also be useful, in case of an address
change, to do one or both of the following:
+ propagate the new address to the parent user/peer descriptor so that new
dialogs will use the correct address from the beginning.
This is trivial to implement, I am just waiting for feedback on this.
+ re-issue a request in case of an address change. This a lot less trivial,
maybe unnecessary, and probably covered by the previous item.
I would seriously consider this patch for addition to 1.4 and 1.2.
The code is very little intrusive, and it would solve in a correct
way the nat traversal problems for which externip/externaddr/stunaddr
are only a partial and expensive workaround.
................
r77630 | rizzo | 2007-07-28 03:29:22 -0600 (Sat, 28 Jul 2007) | 2 lines
simplify a conditional expression using S_OR
................
r77631 | rizzo | 2007-07-28 03:32:10 -0600 (Sat, 28 Jul 2007) | 2 lines
remove an unused string
................
r77647 | rizzo | 2007-07-28 10:25:25 -0600 (Sat, 28 Jul 2007) | 11 lines
start introducing hooks for reference counts on dialog descriptors.
This commit is, for all practical purposes, a no-op, as it only
introduces the dialog_ref() and dialog_unref() methods, and uses them
in a few places (not all the places where they would be needed).
The goal is to start annotating the code with these calls, so the transition
to a proper container will be easier.
Nothing to backport.
................
r77648 | rizzo | 2007-07-28 10:38:07 -0600 (Sat, 28 Jul 2007) | 2 lines
more dialog_ref()/dialog_unref() calls
................
r77650 | rizzo | 2007-07-28 11:16:24 -0600 (Sat, 28 Jul 2007) | 2 lines
more dialog_ref()/dialog_unref() calls
................
r77653 | rizzo | 2007-07-28 17:43:35 -0600 (Sat, 28 Jul 2007) | 7 lines
add some documentation to auto_congest(), and some
dialog_ref/unref (they are a no-op at the moment).
Also clean a pointer after freeing memory to avoid
dangling references, and write a for() loop in canonical form.
In practice, everything in this commit is a no-op.
................
r77654 | rizzo | 2007-07-28 18:06:35 -0600 (Sat, 28 Jul 2007) | 8 lines
make register_unref() return NULL so it is easy to cleanup
the original pointer while calling the function.
on passing add some comments on one of the places where it
is used, and explain why it is safe there.
again, a no-op for practical purposes.
................
r77668 | rizzo | 2007-07-29 02:19:19 -0600 (Sun, 29 Jul 2007) | 6 lines
more documentation on internal representation of incoming SIP messages.
Remove definitions for now-unused flags, and add references to print
routines for other flags.
................
r77669 | rizzo | 2007-07-29 02:58:10 -0600 (Sun, 29 Jul 2007) | 8 lines
back on cleaning up the usage of flags.
Move together flags used in the same way (e.g. dialog only,
dialog-peer, ...) so it will become easier to deal with them
in a more systematic way.
This is being done in stages so it will be easier to detect
breakage, if any should occur.
................
r77682 | rizzo | 2007-07-29 03:27:30 -0600 (Sun, 29 Jul 2007) | 18 lines
remove bit position from description of SIP_* flags.
use AST_FORMAT_AUDIO_MASK instead of playing with AST_FORMAT_MAX_AUDIO
to determine audio formats.
There is a dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call()
which surely needs fixing, namely:
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
* unclear what to use here. We have global_capabilities, which is
* configured from sip.conf, and sip_tech.capabilities, which is
* hardwired to all audio formats.
*/
The latter is possibly something to backport when fixed.
................
r77684 | rizzo | 2007-07-29 04:13:14 -0600 (Sun, 29 Jul 2007) | 6 lines
build the version of sip_tech with no send_digit_begin
at load time instead of duplicating the initializer.
This should remove the risk of forgetting fields in the
initializer.
................
r77697 | rizzo | 2007-07-29 04:49:32 -0600 (Sun, 29 Jul 2007) | 2 lines
minor simplification of a conditional statement
................
r77711 | rizzo | 2007-07-29 14:01:36 -0600 (Sun, 29 Jul 2007) | 14 lines
Move some global 'flags' to individual variables.
Start putting these variables in a single struct (called 'sip_cfg' for the time
being, but it could as well be 'global' or some other name) so it
is easy, when reading the code, to figure out what they are for.
The downside of using struct fields instead of individual global
variables is that the compiler cannot tell if there are unused fields.
But the advantage of not cluttering the namespace and manilpulating
all these variables at once certainly overcome the disadvantagess.
Nothing to backport, again.
................
r77712 | rizzo | 2007-07-29 14:13:32 -0600 (Sun, 29 Jul 2007) | 3 lines
move RT_FROMCONTACT to a proper sip_peer field.
................
r77725 | rizzo | 2007-07-29 14:28:02 -0600 (Sun, 29 Jul 2007) | 2 lines
move the two remaining peer flags to proper variables.
................
r77726 | rizzo | 2007-07-29 14:55:20 -0600 (Sun, 29 Jul 2007) | 8 lines
use a function, cli_yesno(), to produce the output Yes or No for
CLI lines. This helps maintaining consistency on output, slightly
improves readability, and maybe one day will make it easier to
translate the output in other languages (though i have a hard time
believing that a CLI user who needs 'yes' and 'no' to be translated
can actually figure out what he/she is doing!)
................
r77739 | rizzo | 2007-07-29 15:24:56 -0600 (Sun, 29 Jul 2007) | 19 lines
move some dialog-only flags to proper variables, namely
SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT,
SIP_PAGE2_OUTGOING_CALL
These are seldom used so the diff is relatively small.
Note that 'OUTGOING_CALL' is dangerously similar to another
dialog flag, 'SIP_OUTGOING', so the description will need to
clarify the different meaning of the two.
Also note that the description of NOTEXT is a bit unclear - does
it mean we don't support it, or 'not requested or not supported' ?
On passing fix a comment referring to video instead of text.
Finally, mark with XXX a possibly misleading debugging message.
(maybe the latter is worth backporting).
................
r77753 | rizzo | 2007-07-30 02:07:00 -0600 (Mon, 30 Jul 2007) | 5 lines
rename handle_request() to handle_incoming(), as the former
was misleading - the function deals with all incoming packets, be
them requests or responses.
................
r77766 | rizzo | 2007-07-30 04:55:37 -0600 (Mon, 30 Jul 2007) | 12 lines
minor code rearrangements:
+ place the link field at the beginning of struct sip_pvt,
and not somewhere in the middle;
+ in __sip_reliable_xmit, remove a duplicate assignment, and
put the statements in a more logical order (i.e. first copy
the payload and associated info, then copy arguments from the
caller, then finish initializing the headers...)
nothing to backport.
................
r77769 | file | 2007-07-30 08:53:14 -0600 (Mon, 30 Jul 2007) | 20 lines
Merged revisions 77768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r77768 | file | 2007-07-30 11:51:44 -0300 (Mon, 30 Jul 2007) | 12 lines
Merged revisions 77767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 lines
(closes issue #10334)
Reported by: ramonpeek
Pass through the return value from macro_exec through the MacroIf application.
........
................
................
r77770 | russell | 2007-07-30 09:22:20 -0600 (Mon, 30 Jul 2007) | 2 lines
Resolve some compiler warnings so that I can build under dev mode
................
r77772 | file | 2007-07-30 09:49:30 -0600 (Mon, 30 Jul 2007) | 14 lines
Merged revisions 77771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function.
........
................
r77773 | file | 2007-07-30 10:02:01 -0600 (Mon, 30 Jul 2007) | 2 lines
Minor clean up of app_followme.
................
r77779 | file | 2007-07-30 11:12:58 -0600 (Mon, 30 Jul 2007) | 12 lines
Merged revisions 77778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines
(closes issue #10327)
Reported by: kkiely
Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place.
........
................
r77781 | russell | 2007-07-30 11:31:29 -0600 (Mon, 30 Jul 2007) | 24 lines
Merged revisions 77780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Additional changes by me
Fix some problems in channel_find_locked() which can cause an infinite loop.
The reference to the previous channel is set to NULL in some cases. These changes
ensure that the reference to the previous channel gets restored before needing
it again.
I'm not convinced that the code that is setting it to NULL is really the right
thing to do. However, I am making these changes to fix the obvious problem
and just leaving an XXX comment that it needs a better explanation that what
is there now.
........
................
r77784 | tilghman | 2007-07-30 12:45:09 -0600 (Mon, 30 Jul 2007) | 18 lines
Merged revisions 77783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r77783 | tilghman | 2007-07-30 13:43:55 -0500 (Mon, 30 Jul 2007) | 10 lines
Merged revisions 77782 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) | 2 lines
Revert change in revision 71656, even though it fixed a bug, because many people were depending upon the (broken) behavior.
........
................
................
r77786 | russell | 2007-07-30 12:56:29 -0600 (Mon, 30 Jul 2007) | 11 lines
Merged revisions 77785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines
file and I both committed changes for issue #10301. Remove a duplicated
assignment to restore the original value of the previous channel.
........
................
r77787 | tilghman | 2007-07-30 13:11:28 -0600 (Mon, 30 Jul 2007) | 2 lines
Cleanup of res_agi, ensuring thread safety (closes issue #10288)
................
r77789 | russell | 2007-07-30 13:18:24 -0600 (Mon, 30 Jul 2007) | 18 lines
Merged revisions 77788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines
(closes issue #10279)
Reported by: seanbright
Patches:
res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)
Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.
........
................
r77790 | russell | 2007-07-30 13:31:27 -0600 (Mon, 30 Jul 2007) | 5 lines
Remove an XXX comment noting that it would be nice for a declaration to be
inside of a function. (Yes, it would!) Replace it with a note that explains
why it can't be done using the way that the AST_THREADSTORAGE macro is
currently defined.
................
r77791 | russell | 2007-07-30 13:35:33 -0600 (Mon, 30 Jul 2007) | 7 lines
Improve ast_agi_fdprintf() by using the ast_str() API.
* Use a thread local ast_str for building the string that will be written out
to the console for debug, and to the FD for the AGI itself, instead of allocating
a buffer on the heap every time the function is called.
* Use the information contained within the ast_str to determine how many bytes
need to be written instead of calling strlen().
................
r77792 | russell | 2007-07-30 13:39:52 -0600 (Mon, 30 Jul 2007) | 2 lines
Fix the return value of ast_agi_fdprintf() to include the result from ast_carefulwrite()
................
r77793 | rizzo | 2007-07-30 13:42:25 -0600 (Mon, 30 Jul 2007) | 4 lines
print formats as 0x%x instead of %d in a warning message.
Being bitmasks, it is a lot easier to read this way.
................
r77796 | qwell | 2007-07-30 14:19:13 -0600 (Mon, 30 Jul 2007) | 15 lines
Merged revisions 77795 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10083)
........
r77795 | qwell | 2007-07-30 15:17:08 -0500 (Mon, 30 Jul 2007) | 6 lines
Applications like SayAlpha() should not hang up the channel if you
request an "unknown" character such as a comma.
Instead, skip the character and move on.
Issue 10083, initial patch by jsmith, modified by me.
........
................
r77797 | russell | 2007-07-30 14:21:05 -0600 (Mon, 30 Jul 2007) | 16 lines
Merged revisions 77794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) | 8 lines
Fix an issue that could potentially cause corruption of the global iax frame
queue. In the network_thread() loop, it traverses the list using the
AST_LIST_TRAVERSE_SAFE macro. However, to remove an element of the list within
this loop, it used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I
believe could leave some of the internal variables of the SAFE macro invalid.
Mihai says that he already made this change in his local copy and it didn't help
his VNAK storm issues, but I still think it's wrong. :)
........
................
r77799 | russell | 2007-07-30 14:33:44 -0600 (Mon, 30 Jul 2007) | 2 lines
Explicitly set a variable to 1 instead of using the increment operator.
................
r77800 | russell | 2007-07-30 14:36:18 -0600 (Mon, 30 Jul 2007) | 2 lines
Change another unnecessary use of the increment operator to explicitly set the var to 1
................
r77801 | file | 2007-07-30 14:42:28 -0600 (Mon, 30 Jul 2007) | 2 lines
Add support for call forwarding and timeouts to the dialing API.
................
r77808 | tilghman | 2007-07-30 19:10:47 -0600 (Mon, 30 Jul 2007) | 2 lines
Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
................
r77810 | murf | 2007-07-30 21:32:04 -0600 (Mon, 30 Jul 2007) | 1 line
Discovered in experiments on core files: if you wrap the lock and unlock calls with sip_pvt_lock and sip_pvt_unlock, you lose the tracing info you would normally get via DETECT_DEADLOCKS; so I turn these two functions into macros when DETECT_DEADLOCKS is called. This way, you get meaningful stuff in the file and func slots in the lock_info struct.
................
Modified:
team/group/CDRfix5/ (props changed)
team/group/CDRfix5/CHANGES
team/group/CDRfix5/LICENSE
team/group/CDRfix5/Makefile
team/group/CDRfix5/UPGRADE.txt
team/group/CDRfix5/acinclude.m4
team/group/CDRfix5/apps/app_adsiprog.c
team/group/CDRfix5/apps/app_alarmreceiver.c
team/group/CDRfix5/apps/app_amd.c
team/group/CDRfix5/apps/app_authenticate.c
team/group/CDRfix5/apps/app_chanisavail.c
team/group/CDRfix5/apps/app_channelredirect.c
team/group/CDRfix5/apps/app_chanspy.c
team/group/CDRfix5/apps/app_controlplayback.c
team/group/CDRfix5/apps/app_db.c
team/group/CDRfix5/apps/app_dial.c
team/group/CDRfix5/apps/app_dictate.c
team/group/CDRfix5/apps/app_directory.c
team/group/CDRfix5/apps/app_disa.c
team/group/CDRfix5/apps/app_exec.c
team/group/CDRfix5/apps/app_externalivr.c
team/group/CDRfix5/apps/app_festival.c
team/group/CDRfix5/apps/app_flash.c
team/group/CDRfix5/apps/app_followme.c
team/group/CDRfix5/apps/app_getcpeid.c
team/group/CDRfix5/apps/app_image.c
team/group/CDRfix5/apps/app_macro.c
team/group/CDRfix5/apps/app_meetme.c
team/group/CDRfix5/apps/app_minivm.c
team/group/CDRfix5/apps/app_mixmonitor.c
team/group/CDRfix5/apps/app_morsecode.c
team/group/CDRfix5/apps/app_page.c
team/group/CDRfix5/apps/app_parkandannounce.c
team/group/CDRfix5/apps/app_playback.c
team/group/CDRfix5/apps/app_privacy.c
team/group/CDRfix5/apps/app_queue.c
team/group/CDRfix5/apps/app_read.c
team/group/CDRfix5/apps/app_readfile.c
team/group/CDRfix5/apps/app_record.c
team/group/CDRfix5/apps/app_rpt.c
team/group/CDRfix5/apps/app_sayunixtime.c
team/group/CDRfix5/apps/app_senddtmf.c
team/group/CDRfix5/apps/app_sendtext.c
team/group/CDRfix5/apps/app_skel.c
team/group/CDRfix5/apps/app_sms.c
team/group/CDRfix5/apps/app_softhangup.c
team/group/CDRfix5/apps/app_speech_utils.c
team/group/CDRfix5/apps/app_stack.c
team/group/CDRfix5/apps/app_talkdetect.c
team/group/CDRfix5/apps/app_transfer.c
team/group/CDRfix5/apps/app_url.c
team/group/CDRfix5/apps/app_userevent.c
team/group/CDRfix5/apps/app_verbose.c
team/group/CDRfix5/apps/app_voicemail.c
team/group/CDRfix5/apps/app_waitforring.c
[... 21389 lines stripped ...]
More information about the asterisk-commits
mailing list