[asterisk-commits] file: branch file/ah r62363 - in /team/file/ah:
./ apps/ channels/ channels/m...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Apr 30 07:00:52 MST 2007
Author: file
Date: Mon Apr 30 09:00:51 2007
New Revision: 62363
URL: http://svn.digium.com/view/asterisk?view=rev&rev=62363
Log:
Merged revisions 61324,61374-61375,61378-61379,61410,61428-61429,61460,61478,61522,61539,61557,61575-61576,61597,61599,61618,61642,61646-61647,61649,61652,61657,61659-61661,61667,61671,61675,61677,61679,61682,61684,61689,61691,61695,61698,61702,61706,61708,61760,61764,61766-61767,61773,61775,61782,61784,61788,61800,61806,61864,61876,61915,61960,61962,62006,62039,62096,62140-62141,62172,62175,62219,62242,62264,62267-62268,62292,62295,62297-62298,62300,62332 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r61324 | tilghman | 2007-04-10 20:55:26 -0300 (Tue, 10 Apr 2007) | 2 lines
Issue 6082 - New DTMF event for manager
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r61374 | murf | 2007-04-11 10:41:17 -0300 (Wed, 11 Apr 2007) | 1 line
via 8118, a RealTime upgrade to make RT a complete storage abstraction. The store/destroy mechanisms needed these missing peices.
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r61375 | file | 2007-04-11 11:01:53 -0300 (Wed, 11 Apr 2007) | 2 lines
Remove duplicate prototype declaration. (issue #9517 reported by junky)
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r61378 | murf | 2007-04-11 11:09:26 -0300 (Wed, 11 Apr 2007) | 1 line
via 8119, a patch to allow voicemail data to be stored in RealTime.
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r61379 | russell | 2007-04-11 11:13:08 -0300 (Wed, 11 Apr 2007) | 21 lines
Merged revisions 61377 via svnmerge from
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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61376 via svnmerge from
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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r61410 | russell | 2007-04-11 11:49:07 -0300 (Wed, 11 Apr 2007) | 12 lines
Merged revisions 61407 via svnmerge from
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r61407 | russell | 2007-04-11 09:48:01 -0500 (Wed, 11 Apr 2007) | 4 lines
Add "svgz" to the mimetypes table. (issue #9510, bkruse)
In passing, constify the elements of the mimetypes table.
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r61428 | russell | 2007-04-11 12:13:12 -0300 (Wed, 11 Apr 2007) | 22 lines
Merged revisions 61427 via svnmerge from
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
Merged revisions 61426 via svnmerge from
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515, reported by caio1982, fixed by me)
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r61429 | russell | 2007-04-11 12:25:43 -0300 (Wed, 11 Apr 2007) | 3 lines
Add a minor loop optimization to the custom device state callback. Once the
correct device is found, it should just break out of the loop ...
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r61460 | nadi | 2007-04-11 12:48:54 -0300 (Wed, 11 Apr 2007) | 25 lines
Merged revisions 61342,61372-61373,61443 via svnmerge from
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r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines
AOCD's are now exported to asterisk channel variables.
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r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines
Ignore facility messages in case we don't have a corresponding channel object.
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r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines
Export AOCD variables on misdn_hangup.
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r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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r61478 | russell | 2007-04-11 13:06:37 -0300 (Wed, 11 Apr 2007) | 21 lines
Merged revisions 61477 via svnmerge from
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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61476 via svnmerge from
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488, reported by makoto, fixed by me)
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r61522 | file | 2007-04-11 14:07:17 -0300 (Wed, 11 Apr 2007) | 2 lines
Output verbose messages to the normal logger as well. (issue #9476 reported by gdalgliesh)
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r61539 | dhubbard | 2007-04-11 16:11:32 -0300 (Wed, 11 Apr 2007) | 1 line
added option_minmemfree for use in asterisk.conf to specify the amount of minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information
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r61557 | file | 2007-04-11 17:21:18 -0300 (Wed, 11 Apr 2007) | 2 lines
Add a configure script check for sysinfo support.
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r61575 | dhubbard | 2007-04-11 17:59:08 -0300 (Wed, 11 Apr 2007) | 1 line
added HAVE_SYSINFO preprocessor directives for portability and general happiness
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r61576 | dhubbard | 2007-04-11 18:13:44 -0300 (Wed, 11 Apr 2007) | 1 line
changed #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO)
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r61597 | dhubbard | 2007-04-11 19:10:26 -0300 (Wed, 11 Apr 2007) | 1 line
fixed the '-e' command line option for minmemfree. updated doc/asterisk-conf.tex
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r61599 | dhubbard | 2007-04-11 19:19:14 -0300 (Wed, 11 Apr 2007) | 1 line
clarified 'minmemfree' description in doc/asterisk-conf.tex
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r61618 | file | 2007-04-12 16:32:00 -0300 (Thu, 12 Apr 2007) | 2 lines
Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee)
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r61642 | file | 2007-04-13 13:35:33 -0300 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61641 via svnmerge from
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r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines
Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)
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r61646 | murf | 2007-04-13 14:11:53 -0300 (Fri, 13 Apr 2007) | 9 lines
Merged revisions 61644 via svnmerge from
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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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r61647 | russell | 2007-04-13 14:15:45 -0300 (Fri, 13 Apr 2007) | 11 lines
Merged revisions 61645 via svnmerge from
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r61645 | russell | 2007-04-13 12:10:19 -0500 (Fri, 13 Apr 2007) | 3 lines
Eliminate a compiler warning with ODBC_STORAGE enabled so that it will build
under dev-mode.
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r61649 | file | 2007-04-13 14:21:53 -0300 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61648 via svnmerge from
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r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
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r61652 | file | 2007-04-13 15:09:29 -0300 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61651 via svnmerge from
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r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr 2007) | 2 lines
Do not bother looking for a result if none are present.
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r61657 | file | 2007-04-13 16:18:46 -0300 (Fri, 13 Apr 2007) | 18 lines
Merged revisions 61656 via svnmerge from
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r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61655 via svnmerge from
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r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines
Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140)
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r61659 | murf | 2007-04-13 18:22:01 -0300 (Fri, 13 Apr 2007) | 9 lines
Merged revisions 61658 via svnmerge from
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r61658 | murf | 2007-04-13 15:17:20 -0600 (Fri, 13 Apr 2007) | 1 line
This is a fix to the way CDR merge handles the data that results from ForkCDR.
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r61660 | dhubbard | 2007-04-13 18:23:10 -0300 (Fri, 13 Apr 2007) | 1 line
added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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r61661 | junky | 2007-04-14 15:22:39 -0300 (Sat, 14 Apr 2007) | 3 lines
test my new trunk access ;)
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r61667 | oej | 2007-04-16 12:40:32 -0300 (Mon, 16 Apr 2007) | 2 lines
Doxygen changes
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r61671 | oej | 2007-04-18 04:57:18 -0300 (Wed, 18 Apr 2007) | 7 lines
Mini-voicemail - an embryo for a new voicemail system based on building
blocks instead of one large monolithic app. Supports multiple templates
and is designed mostly for voicemail delivery over e-mail.
There's a todo with a list of ideas in the source code if you want
to contribute. Feedback is appreciated!
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r61675 | oej | 2007-04-18 17:39:31 -0300 (Wed, 18 Apr 2007) | 10 lines
Merged revisions 61674 via svnmerge from
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r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines
Issue #9554 - Improve RTCP (Dave Troy)
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r61677 | oej | 2007-04-18 17:48:13 -0300 (Wed, 18 Apr 2007) | 10 lines
Merged revisions 61676 via svnmerge from
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r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines
Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin!
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r61679 | kpfleming | 2007-04-18 19:11:02 -0300 (Wed, 18 Apr 2007) | 10 lines
Merged revisions 61678 via svnmerge from
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r61678 | kpfleming | 2007-04-18 17:10:23 -0500 (Wed, 18 Apr 2007) | 2 lines
allow external build systems to extract the required sound file versions
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r61682 | tilghman | 2007-04-18 23:51:21 -0300 (Wed, 18 Apr 2007) | 21 lines
Merged revisions 61681 via svnmerge from
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r61681 | tilghman | 2007-04-18 21:45:05 -0500 (Wed, 18 Apr 2007) | 13 lines
Merged revisions 61680 via svnmerge from
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r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) | 5 lines
Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can
cause Asterisk to crash. The reason this needs to be fixed in the functions
instead of in AMI is because Channel can legitimately be NULL, such as when
retrieving global variables.
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r61684 | tilghman | 2007-04-19 01:37:29 -0300 (Thu, 19 Apr 2007) | 10 lines
Merged revisions 61683 via svnmerge from
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r61683 | tilghman | 2007-04-18 23:36:20 -0500 (Wed, 18 Apr 2007) | 2 lines
Bug 9557 - simple reason why reading a function always returned NULL
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r61689 | oej | 2007-04-20 05:41:24 -0300 (Fri, 20 Apr 2007) | 4 lines
Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.
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r61691 | russell | 2007-04-20 15:23:24 -0300 (Fri, 20 Apr 2007) | 12 lines
Merged revisions 61690 via svnmerge from
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r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) | 4 lines
Fix the UpdateConfig manager action to properly treat "variables" and "objects"
differently (a=b versus a=>b).
(issue #9568, reported by pari, patch by me)
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r61695 | qwell | 2007-04-20 16:54:54 -0300 (Fri, 20 Apr 2007) | 21 lines
Merged revisions 61694 via svnmerge from
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r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri, 20 Apr 2007) | 13 lines
Merged revisions 61692 via svnmerge from
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r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 lines
If the '* to hangup' option is not enabled, we don't need to disable * as a valid exit key.
If it was enabled, this statement would've never been checked in the first place.
Issue #9552
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r61698 | russell | 2007-04-20 17:43:05 -0300 (Fri, 20 Apr 2007) | 10 lines
Merged revisions 61697 via svnmerge from
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r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines
Remove a stray debug message introduced by a recent commit.
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r61702 | russell | 2007-04-20 18:12:53 -0300 (Fri, 20 Apr 2007) | 8 lines
Merge changes from team/russell/iax2_osp
This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531, reported by and original patch by homesick, patch updated by me)
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r61706 | qwell | 2007-04-20 18:16:14 -0300 (Fri, 20 Apr 2007) | 20 lines
Merged revisions 61705 via svnmerge from
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r61705 | qwell | 2007-04-20 16:15:29 -0500 (Fri, 20 Apr 2007) | 12 lines
Merged revisions 61704 via svnmerge from
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r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 lines
Fix an issue that I noticed while looking over issue 9571.
The reload timestamp was getting set after reloading the built-in stuff, and before the modules.
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r61708 | qwell | 2007-04-20 18:37:04 -0300 (Fri, 20 Apr 2007) | 16 lines
Merged revisions 61707 via svnmerge from
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r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines
Avoid invalid seqno cycling detection.
Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
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r61760 | russell | 2007-04-23 12:34:51 -0300 (Mon, 23 Apr 2007) | 3 lines
Add OSP support for IAX2 to the changes file. Also, slightly reorganize some
of the content.
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r61764 | russell | 2007-04-23 14:58:15 -0300 (Mon, 23 Apr 2007) | 12 lines
Merged revisions 61763 via svnmerge from
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r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | 4 lines
Ensure that digits passing through Asterisk have a reasonable minimum length.
It is currently 100 ms. If someone thinks this should be different, feel free
to speak up. (related to issues #8944, #9250, and #9348)
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r61766 | russell | 2007-04-23 15:19:42 -0300 (Mon, 23 Apr 2007) | 13 lines
Merged revisions 61765 via svnmerge from
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r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | 5 lines
Some dialplan functions, such as CUT(), expect to operate on variables on a
channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
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r61767 | russell | 2007-04-23 15:49:19 -0300 (Mon, 23 Apr 2007) | 3 lines
When building a JSON encoded string in the GetConfigJSON manager action, escape
the '\' and '"' characters. (issue #9475, reported by pari, patch by me)
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r61773 | file | 2007-04-24 13:10:10 -0300 (Tue, 24 Apr 2007) | 18 lines
Merged revisions 61772 via svnmerge from
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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines
Merged revisions 61771 via svnmerge from
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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r61775 | russell | 2007-04-24 13:17:36 -0300 (Tue, 24 Apr 2007) | 13 lines
Merged revisions 61774 via svnmerge from
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r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
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r61782 | russell | 2007-04-24 16:03:16 -0300 (Tue, 24 Apr 2007) | 14 lines
Merged revisions 61781 via svnmerge from
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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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r61784 | dhubbard | 2007-04-24 16:08:28 -0300 (Tue, 24 Apr 2007) | 2 lines
removed #if 0 block from chan_zap restart_monitor()
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r61788 | russell | 2007-04-24 18:37:00 -0300 (Tue, 24 Apr 2007) | 20 lines
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r61787 | russell | 2007-04-24 16:34:53 -0500 (Tue, 24 Apr 2007) | 12 lines
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r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines
Don't crash if a manager connection provides a username that exists in
manager.conf but does not have a password, and also requests MD5
authentication. (ASA-2007-012)
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r61800 | russell | 2007-04-25 13:23:00 -0300 (Wed, 25 Apr 2007) | 19 lines
Merged revisions 61799 via svnmerge from
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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines
Merged revisions 61798 via svnmerge from
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587, reported and patched by xrg)
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r61806 | file | 2007-04-25 16:27:42 -0300 (Wed, 25 Apr 2007) | 18 lines
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r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines
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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines
Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
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r61864 | russell | 2007-04-25 18:15:19 -0300 (Wed, 25 Apr 2007) | 18 lines
Merged revisions 61863 via svnmerge from
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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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r61876 | russell | 2007-04-25 19:01:37 -0300 (Wed, 25 Apr 2007) | 18 lines
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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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r61915 | kpfleming | 2007-04-25 19:34:58 -0300 (Wed, 25 Apr 2007) | 18 lines
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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286)
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r61960 | file | 2007-04-25 22:29:23 -0300 (Wed, 25 Apr 2007) | 18 lines
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61959 | file | 2007-04-25 21:27:18 -0400 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 lines
Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn)
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r61962 | file | 2007-04-25 22:50:02 -0300 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2 lines
Don't always say that the channel is being paused if it is actually being unpaused in the Manager ack message. (reported by jsmith in #asterisk-bugs)
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r62006 | russell | 2007-04-26 00:24:01 -0300 (Thu, 26 Apr 2007) | 10 lines
Merged revisions 62005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines
Missed an ast_app_group_discard during merge. Thanks blitzrage!
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r62039 | file | 2007-04-26 13:35:14 -0300 (Thu, 26 Apr 2007) | 18 lines
Merged revisions 62038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines
Merged revisions 62037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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r62096 | oej | 2007-04-27 05:22:41 -0300 (Fri, 27 Apr 2007) | 2 lines
Blocking patch to 1.4 that was alredy in trunk
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r62140 | oej | 2007-04-27 11:37:10 -0300 (Fri, 27 Apr 2007) | 20 lines
Merged revisions 62137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
Merged revisions 62126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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r62141 | oej | 2007-04-27 11:40:28 -0300 (Fri, 27 Apr 2007) | 2 lines
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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r62172 | russell | 2007-04-27 13:15:47 -0300 (Fri, 27 Apr 2007) | 14 lines
Merged revisions 62171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
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r62175 | qwell | 2007-04-27 13:18:51 -0300 (Fri, 27 Apr 2007) | 19 lines
Merged revisions 62174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, 27 Apr 2007) | 11 lines
Merged revisions 62173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines
This transcoder message needn't be a NOTICE.
I've seen it cause confusion more than a few times.
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r62219 | russell | 2007-04-27 18:11:46 -0300 (Fri, 27 Apr 2007) | 19 lines
Merged revisions 62218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)
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r62242 | russell | 2007-04-27 19:08:54 -0300 (Fri, 27 Apr 2007) | 5 lines
Add a min-announce-frequency option to queues.conf which allows you to control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
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r62264 | russell | 2007-04-28 16:23:46 -0300 (Sat, 28 Apr 2007) | 2 lines
Remove a message that goes to LOG_ERROR that's not really an error.
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r62267 | russell | 2007-04-28 16:52:37 -0300 (Sat, 28 Apr 2007) | 7 lines
Merge changes from team/russell/dundi_results
This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.
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r62268 | russell | 2007-04-28 16:53:12 -0300 (Sat, 28 Apr 2007) | 3 lines
Update the DUNDi section of the documentation with example usage of DUNDIQUERY
and DUNDIRESULT. Also, update the automatically generated application docs.
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r62292 | russell | 2007-04-28 18:01:44 -0300 (Sat, 28 Apr 2007) | 19 lines
Merge changes from team/russell/events
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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r62295 | russell | 2007-04-28 18:26:00 -0300 (Sat, 28 Apr 2007) | 2 lines
Reformat some of iax2.h and convert comments to doxygen format
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r62297 | russell | 2007-04-28 18:54:44 -0300 (Sat, 28 Apr 2007) | 2 lines
Enable the functionality of the 'o' option to "optimize talker" by default.
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r62298 | russell | 2007-04-28 18:55:00 -0300 (Sat, 28 Apr 2007) | 1 line
note MeetMe change in CHANGES
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r62300 | russell | 2007-04-28 18:56:46 -0300 (Sat, 28 Apr 2007) | 9 lines
Blocked revisions 62299 via svnmerge
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r62299 | russell | 2007-04-28 16:56:20 -0500 (Sat, 28 Apr 2007) | 2 lines
Note that the "talker optimization" option will be enabled by default in 1.6
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r62332 | russell | 2007-04-29 02:51:18 -0300 (Sun, 29 Apr 2007) | 11 lines
Merged revisions 62331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines
Fix a bug that made the "language" setting in zapata.conf not
functional. (issue #9626, reported and fixed by sergee)
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Added:
team/file/ah/apps/app_minivm.c
- copied unchanged from r62332, trunk/apps/app_minivm.c
team/file/ah/configs/extensions_minivm.conf.sample
- copied unchanged from r62332, trunk/configs/extensions_minivm.conf.sample
team/file/ah/configs/minivm.conf.sample
- copied unchanged from r62332, trunk/configs/minivm.conf.sample
team/file/ah/include/asterisk/event.h
- copied unchanged from r62332, trunk/include/asterisk/event.h
team/file/ah/include/asterisk/event_defs.h
- copied unchanged from r62332, trunk/include/asterisk/event_defs.h
team/file/ah/main/event.c
- copied unchanged from r62332, trunk/main/event.c
team/file/ah/res/res_eventtest.c
- copied unchanged from r62332, trunk/res/res_eventtest.c
Modified:
team/file/ah/ (props changed)
team/file/ah/CHANGES
team/file/ah/UPGRADE.txt
team/file/ah/apps/app_dial.c
team/file/ah/apps/app_meetme.c
team/file/ah/apps/app_queue.c
team/file/ah/apps/app_speech_utils.c
team/file/ah/apps/app_voicemail.c
team/file/ah/channels/chan_agent.c
team/file/ah/channels/chan_iax2.c
team/file/ah/channels/chan_mgcp.c
team/file/ah/channels/chan_misdn.c
team/file/ah/channels/chan_oss.c
team/file/ah/channels/chan_sip.c
team/file/ah/channels/chan_zap.c
team/file/ah/channels/iax2-parser.c
team/file/ah/channels/iax2-parser.h
team/file/ah/channels/iax2.h
team/file/ah/channels/misdn/isdn_lib.c
team/file/ah/channels/misdn/isdn_lib.h
team/file/ah/codecs/codec_zap.c
team/file/ah/configs/manager.conf.sample
team/file/ah/configs/queues.conf.sample
team/file/ah/configs/sip.conf.sample
team/file/ah/configs/voicemail.conf.sample
team/file/ah/configure
team/file/ah/configure.ac
team/file/ah/doc/ast_appdocs.tex
team/file/ah/doc/asterisk-conf.tex
team/file/ah/doc/dundi.tex
team/file/ah/funcs/func_callerid.c
team/file/ah/funcs/func_cdr.c
team/file/ah/funcs/func_channel.c
team/file/ah/funcs/func_devstate.c
team/file/ah/funcs/func_groupcount.c
team/file/ah/funcs/func_timeout.c
team/file/ah/include/asterisk.h
team/file/ah/include/asterisk/app.h
team/file/ah/include/asterisk/autoconfig.h.in
team/file/ah/include/asterisk/channel.h
team/file/ah/include/asterisk/config.h
team/file/ah/include/asterisk/lock.h
team/file/ah/include/asterisk/manager.h
team/file/ah/include/asterisk/options.h
team/file/ah/include/asterisk/rtp.h
team/file/ah/main/Makefile
team/file/ah/main/app.c
team/file/ah/main/asterisk.c
team/file/ah/main/cdr.c
team/file/ah/main/channel.c
team/file/ah/main/cli.c
team/file/ah/main/config.c
team/file/ah/main/dial.c
team/file/ah/main/http.c
team/file/ah/main/loader.c
team/file/ah/main/logger.c
team/file/ah/main/manager.c
team/file/ah/main/pbx.c
team/file/ah/main/rtp.c
team/file/ah/main/say.c
team/file/ah/pbx/pbx_dundi.c
team/file/ah/res/res_monitor.c
team/file/ah/sounds/Makefile
Propchange: team/file/ah/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/file/ah/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/file/ah/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Apr 30 09:00:51 2007
@@ -1,1 +1,1 @@
-/trunk:1-61306
+/trunk:1-62362
Modified: team/file/ah/CHANGES
URL: http://svn.digium.com/view/asterisk/team/file/ah/CHANGES?view=diff&rev=62363&r1=62362&r2=62363
==============================================================================
--- team/file/ah/CHANGES (original)
+++ team/file/ah/CHANGES Mon Apr 30 09:00:51 2007
@@ -1,73 +1,6 @@
-------------------------------------------------------------------------------
--- Functionality changes since Asterisk 1.4-beta was branched ----------------
-------------------------------------------------------------------------------
-
-Miscellaneous
--------------
-
- * Added the bindaddr option to gtalk.conf.
- * Argument support for Gosub application
- * Ability to set process limits without restarting Asterisk
- * SS7 support in chan_zap (via libss7 library)
- * Proper codec support in chan_skinny.
- * AEL upgraded to use the Gosub with Arguments instead
- of Macro application, to hopefully reduce the problems
- seen with the artificially low stack ceiling that
- Macro bumps into. Macros can only call other Macros
- to a depth of 7. Tests run using gosub, show depths
- limited only by virtual memory. A small test demonstrated
- recursive call depths of 100,000 without problems.
- * Ability to use libcap to set high ToS bits when non-root
- on Linux. If configure is unable to find libcap then you
- can use --with-cap to specify the path.
- * H323 remote hold notification support added (by NOTIFY message
- and/or H.450 supplementary service)
- * Added keepstats option to queues.conf which will keep queue
- statistics during a reload.
- * Added rotatetimestamp option to logger.conf which will use
- the time to name the logger files instead of sequence number.
- * setinterfacevar option in queues.conf also now sets a variable
- called MEMBERNAME which contains the member's name.
- * Added Masquerade manager event for when a masquerade happens between
- two channels.
- * Added 'Strategy' field to manager event QueueParams which represents
- the queue strategy in use.
- * From the to-do lists: straighten out the app timeout args:
- Wait() app now really does 0.3 seconds- was truncating arg to an int.
- WaitExten() same as Wait().
- Congestion() - Now takes floating pt. argument.
- Busy() - now takes floating pt. argument.
- Read() - timeout now can be floating pt.
- WaitForRing() now takes floating pt timeout arg.
- SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
- * Added 'C' option to Meetme which causes a caller to continue in the dialplan
- when kicked out.
- * Added option to run macro when a queue member is connected to a caller,
- see queues.conf.sample for details.
- * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
- setqueueentryvar options for each queue, see queues.conf.sample for details.
- * Brazilian Portuguese (pt-BR) in VM, and say.c was added via patch from cfassoni.
- * CID matching information is now shown when doing 'dialplan show'.
- * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
- does not count paused queue members as unavailable.
- * Added maxfiles option to options section of asterisk.conf which allows you to specify
- what Asterisk should set as the maximum number of open files when it loads.
- * Added the jittertargetextra configuration option.
- * Added the trunkmaxsize configuration option to chan_iax2.
- * Added G729 passthrough support to chan_phone for Sigma Designs boards.
- * Added the parkedcalltransfers option to features.conf
- * Added 's' option to Page application.
- * Added the srvlookup option to iax.conf
- * Added 'E' and 'V' commands to ExternalIVR.
- * Added 'DBDel' and 'DBDelTree' manager commands.
- * Added 'o' and 'X' options to Chanspy.
- * Added the parkedcallreparking option to features.conf
- * SMDI is now enabled in voicemail using the smdienable option.
- * Added zap show version CLI command to chan_zap.
- * Added a new CDR module, cdr_sqlite3_custom.
- * Added a new realtime configuration module, res_config_sqlite
- * Added a new dialplan application, Bridge, which allows you to bridge the
- calling channel to any other active channel on the system.
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
@@ -77,14 +10,14 @@
* enable https support for builtin web server.
See configs/http.conf.sample for details.
* Added a new action, GetConfigJSON, which can return the contents of an
- Asterisk configuration file in JSON format. This is intended to help
- improve the performance of AJAX applications using the manager interface
- over HTTP.
+ Asterisk configuration file in JSON format. This is intended to help
+ improve the performance of AJAX applications using the manager interface
+ over HTTP.
* SIP and IAX manager events now use "ChannelType" in all cases where we
- indicate channel driver. Previously, we used a mixture of "Channel"
[... 18357 lines stripped ...]
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