[asterisk-commits] phsultan: branch phsultan/jabberreceive r169482 - in /team/phsultan/jabberrec...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 20 04:40:23 CST 2009
Author: phsultan
Date: Tue Jan 20 04:40:23 2009
New Revision: 169482
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=169482
Log:
Merged revisions 168265,168269-168270,168334,168479,168481,168485,168497,168508,168517,168522-168523,168526,168539,168547,168562,168575,168578-168579,168585,168588,168591,168594,168599,168601,168604,168609-168610,168613,168615,168619,168623,168626,168629,168636,168638-168639,168705,168711-168712,168719,168722,168725,168728,168732,168734,168737,168746,168759-168760,168832,168898,168941,168976,169044,169080,169116,169153,169211,169277,169325,169327,169365,169367,169369,169438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r168265 | mvanbaak | 2009-01-10 00:04:46 +0100 (Sat, 10 Jan 2009) | 9 lines
Add a script to find out the correct settings for Asterisk behind NAT
(closes issue #13065)
Reported by: tzafrir
Patches:
sip_nat_settings uploaded by tzafrir (license 46)
sip_nat_settings_6 uploaded by mvanbaak (license 7)
Tested by: tzafrir, pabelanger, Dovid and moi
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r168269 | rmudgett | 2009-01-10 00:15:26 +0100 (Sat, 10 Jan 2009) | 1 line
Spacing change
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r168270 | kpfleming | 2009-01-10 00:16:08 +0100 (Sat, 10 Jan 2009) | 9 lines
Merged revisions 168267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line
update to use new sound file packages that include license files
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r168334 | tilghman | 2009-01-10 02:42:45 +0100 (Sat, 10 Jan 2009) | 2 lines
sizeof for a stringfield is 4. Kinda low for reconstructing a field value.
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r168479 | oej | 2009-01-12 15:35:09 +0100 (Mon, 12 Jan 2009) | 2 lines
Don't include swap.h unless we have swapctl
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r168481 | russell | 2009-01-12 15:57:49 +0100 (Mon, 12 Jan 2009) | 10 lines
Merged revisions 168480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines
s/ringdance/ringcadence/ for Bulgaria
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r168485 | mmichelson | 2009-01-12 16:00:00 +0100 (Mon, 12 Jan 2009) | 13 lines
Merged revisions 168482 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines
I am reverting the fix made in revision 168128 (and its upward merges)
after being contacted by Olle Johansson and being shown how this fix is
incorrect. Thanks to Olle for clearing this up for me.
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r168497 | oej | 2009-01-12 17:31:27 +0100 (Mon, 12 Jan 2009) | 2 lines
Better to use the proper app name
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r168508 | jpeeler | 2009-01-12 21:53:04 +0100 (Mon, 12 Jan 2009) | 15 lines
Merged revisions 168507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
(closes issue #12269)
Reported by: IgorG
Tested by: denisgalvao
This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
Review: http://reviewboard.digium.com/r/35/
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r168517 | jpeeler | 2009-01-12 22:51:46 +0100 (Mon, 12 Jan 2009) | 12 lines
Merged revisions 168516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines
(closes issue #13881)
Reported by: hoowa
Update the app CDR field for AGI commands that are not executing an application via "exec".
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r168522 | tilghman | 2009-01-13 00:06:12 +0100 (Tue, 13 Jan 2009) | 3 lines
Some platforms (notably, the BSDs) have a more efficient implementation called
closefrom(3).
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r168523 | mmichelson | 2009-01-13 00:12:30 +0100 (Tue, 13 Jan 2009) | 11 lines
bump the verbosity of a message in srv.c up by one. It used to be
at this level prior to a large patch merge which converted ast_verbose
calls to ast_verb
(closes issue #14221)
Reported by: jcovert
Patches:
srv.c.patch uploaded by jcovert (license 551)
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r168526 | tilghman | 2009-01-13 00:45:51 +0100 (Tue, 13 Jan 2009) | 12 lines
Merged revisions 167095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines
Repeat attempts to write when we receive -EAGAIN from the driver, as detailed
in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
Reported by: Jerry Geis (via the -users list)
Fixed by: me (license 14)
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r168539 | dhubbard | 2009-01-13 17:02:13 +0100 (Tue, 13 Jan 2009) | 1 line
correct a CLI description
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r168547 | tilghman | 2009-01-13 18:51:12 +0100 (Tue, 13 Jan 2009) | 13 lines
Merged revisions 168546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines
If either conditional is NULL, don't try copying it.
(closes issue #14226)
Reported by: caspy
Patches:
20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
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r168562 | russell | 2009-01-13 20:22:13 +0100 (Tue, 13 Jan 2009) | 10 lines
Merged revisions 168561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines
Revert unnecessary indications API change from rev 122314
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r168575 | mmichelson | 2009-01-13 22:18:13 +0100 (Tue, 13 Jan 2009) | 13 lines
Allow specifying a port number in the user portion of a register => line in sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
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r168578 | twilson | 2009-01-13 23:22:34 +0100 (Tue, 13 Jan 2009) | 14 lines
Merged revisions 168551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines
Don't pass a value with a side effect to a macro
(closes issue #14176)
Reported by: paraeco
Patches:
chan_sip.c.diff uploaded by paraeco (license 658)
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r168579 | mmichelson | 2009-01-13 23:30:59 +0100 (Tue, 13 Jan 2009) | 13 lines
Clarify a message that app_queue prints and change to a debug-level message
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
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r168585 | twilson | 2009-01-14 00:00:27 +0100 (Wed, 14 Jan 2009) | 8 lines
Add option to hide console connect messages
(closes issue #14222)
Reported by: jamesgolovich
Patches:
asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176)
Tested by: otherwiseguy
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r168588 | twilson | 2009-01-14 00:05:43 +0100 (Wed, 14 Jan 2009) | 5 lines
Fully overwrite a same-named file when uploading
(closes issue #14190)
Reported by: timking
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r168591 | tilghman | 2009-01-14 00:57:46 +0100 (Wed, 14 Jan 2009) | 6 lines
Janitor patch for chan_misdn (make channel variable access safe)
(closes issue #12887)
Reported by: pputman
Patches:
chan_misdn_threadsafe.patch uploaded by pputman (license 81)
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r168594 | twilson | 2009-01-14 03:00:40 +0100 (Wed, 14 Jan 2009) | 27 lines
Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
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r168599 | mmichelson | 2009-01-14 17:20:37 +0100 (Wed, 14 Jan 2009) | 15 lines
Blocked revisions 168598 via svnmerge
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r168598 | mmichelson | 2009-01-14 10:19:26 -0600 (Wed, 14 Jan 2009) | 8 lines
Fix a logic error I found while searching through chan_agent.c
I found that the allow_multiple_logins function would never return
0 due to an incorrect comparison being used when traversing the
list of agents. While I was modifying this function, I also did
a little bit of coding guidelines cleanup, too.
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r168601 | tilghman | 2009-01-14 19:27:57 +0100 (Wed, 14 Jan 2009) | 2 lines
Mostly spacing changes; no functionality change at all.
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r168604 | tilghman | 2009-01-14 20:11:14 +0100 (Wed, 14 Jan 2009) | 14 lines
Merged revisions 168603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines
Don't read into a buffer without first checking if a value is beyond the end.
(closes issue #13600)
Reported by: atis
Patches:
20090106__bug13600.diff.txt uploaded by Corydon76 (license 14)
Tested by: atis
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r168609 | mvanbaak | 2009-01-14 20:36:57 +0100 (Wed, 14 Jan 2009) | 13 lines
Fix compilation on FreeBSD and OSX
This started as work to fix the 'core show sysinfo'
CLI command but while working on it oej
pointed out that read_credentials did not compile neither.
So while being there, fix that as well.
Thanks for all the testing oej!
(closes issue #14129)
Reported by: ys
Tested by: oej, mvanbaak
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r168610 | mmichelson | 2009-01-14 21:13:48 +0100 (Wed, 14 Jan 2009) | 9 lines
Restore the "sip show users" and "sip show user" CLI commands
(closes issue #14180)
Reported by: amorsen
Patches:
sip_show_users_161v3.diff uploaded by putnopvut (license 60)
Tested by: blitzrage, amorsen
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r168613 | murf | 2009-01-14 21:51:26 +0100 (Wed, 14 Jan 2009) | 9 lines
Merged revisions 168608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line
app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning.
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r168615 | seanbright | 2009-01-14 21:58:26 +0100 (Wed, 14 Jan 2009) | 16 lines
Merged revisions 168614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines
Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and
be sure to run dahdi_test in 1.6.x and trunk instead of zttest.
(closes issue #14132)
Reported by: dsedivec
Patches:
asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638)
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r168619 | dbailey | 2009-01-14 22:19:45 +0100 (Wed, 14 Jan 2009) | 8 lines
This fixes a problem where MWI FSK spills were being injected onto off hook fxs lines.
(closes issue #14143)
Reported by: alecdavis
Patches:
chan_dahdi-14143.patch.txt uploaded by dbailey (license )
Tested by: alecdavis
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r168623 | rmudgett | 2009-01-14 22:51:06 +0100 (Wed, 14 Jan 2009) | 11 lines
Merged revisions 168622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) | 4 lines
* Fixed create_process() allocation of process ID values.
The allocated process IDs could overflow their respective
NT and TE fields. Affects outgoing calls.
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r168626 | seanbright | 2009-01-15 00:10:48 +0100 (Thu, 15 Jan 2009) | 7 lines
Don't crash when typing 'core set verbose' or 'core set debug' by themselves.
(closes issue #14219)
Reported by: jamesgolovich
Patches:
asterisk-setverbosecrash.diff.txt uploaded by jamesgolovich (license 176)
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r168629 | mmichelson | 2009-01-15 01:14:17 +0100 (Thu, 15 Jan 2009) | 24 lines
Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
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r168636 | oej | 2009-01-15 14:01:52 +0100 (Thu, 15 Jan 2009) | 5 lines
Add support for setting the Reason header when cancelling a call in the queue
because someone else answered. Previously, only dial() was supported.
EDV-102
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r168638 | oej | 2009-01-15 14:35:50 +0100 (Thu, 15 Jan 2009) | 8 lines
Add capability to remove added SIP headers *before* INVITE is generated.
(closes issue #14246)
Reported by: klaus3000
Patches:
2patch_chan_sip_SIPRemoveHeader_trunk.txt uploaded by klaus3000 (license 65)
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r168639 | oej | 2009-01-15 14:37:46 +0100 (Thu, 15 Jan 2009) | 3 lines
Related to issue #14246
Update changes for SIPRemoveHeader()
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r168705 | seanbright | 2009-01-15 16:33:18 +0100 (Thu, 15 Jan 2009) | 11 lines
Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
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r168711 | oej | 2009-01-15 18:55:53 +0100 (Thu, 15 Jan 2009) | 4 lines
Clarify some misunderstandings and make it even more clear that you can refer to a peer
in the register= line.
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r168712 | oej | 2009-01-15 19:08:59 +0100 (Thu, 15 Jan 2009) | 3 lines
Make sure that we have the same terminology in sip.conf.sample and the source code warning.
Thanks Nick Lewis for pointing this out in the bug tracker.
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r168719 | tilghman | 2009-01-15 19:39:56 +0100 (Thu, 15 Jan 2009) | 4 lines
Resolve issue with negative vs non-negative length parameters.
(closes issue #14245)
Reported by: dveiga
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r168722 | oej | 2009-01-15 19:47:14 +0100 (Thu, 15 Jan 2009) | 10 lines
Merged revisions 168721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines
Meetme actually has realtime but wasn't documented
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r168725 | mmichelson | 2009-01-15 20:00:06 +0100 (Thu, 15 Jan 2009) | 17 lines
Remove an unneeded condition for line addition to a SIP request/response
In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
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r168728 | mmichelson | 2009-01-15 20:16:29 +0100 (Thu, 15 Jan 2009) | 3 lines
Fix the compactheaders option in sip.conf
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r168732 | mmichelson | 2009-01-15 21:00:46 +0100 (Thu, 15 Jan 2009) | 3 lines
Add missing brace
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r168734 | kpfleming | 2009-01-15 21:18:53 +0100 (Thu, 15 Jan 2009) | 5 lines
remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC
this stops modules from being linked against both sets of libraries on systems that have both installed
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r168737 | murf | 2009-01-15 21:54:59 +0100 (Thu, 15 Jan 2009) | 16 lines
This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff.
If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld);
then it won't complain about the empty arg (c,,...) and fabled's patch
won't let it swap the commas for pipes.
Ran it thru my dialplan and no complaints.
(closes issue #14169)
Reported by: fabled
Patches:
function-argument-separator-fix.diff uploaded by fabled (license 448)
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r168746 | murf | 2009-01-16 01:34:31 +0100 (Fri, 16 Jan 2009) | 20 lines
Merged revisions 168745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines
This patch fixes a problem where a goto (or jump, in this case)
fails a consistency check because it can't find a matching
extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern
into a regex and uses the regex code to determine the match.
I tested using the AEL code the user supplied, and now,
the consistency check passes.
(closes issue #14141)
Reported by: dimas
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r168759 | russell | 2009-01-16 17:18:41 +0100 (Fri, 16 Jan 2009) | 1 line
build in dev mode
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r168760 | russell | 2009-01-16 18:09:13 +0100 (Fri, 16 Jan 2009) | 2 lines
Fix a spelling mistake.
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r168832 | tilghman | 2009-01-16 19:49:09 +0100 (Fri, 16 Jan 2009) | 13 lines
Merged revisions 168828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines
Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475)
Reported by: chappell
Patches:
vm_multilang.patch uploaded by chappell (license 8)
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r168898 | mmichelson | 2009-01-16 20:54:39 +0100 (Fri, 16 Jan 2009) | 26 lines
Fix a logic error that occur when using the timerfd interface
This sequence of events posed a problem
timerfd_timer_open
timerfd_timer_enable_continuous
timerfd_timer_set_rate
timerfd_timer_disable_continuous
The reason was that the timing module was written under the assumption
that timerfd_timer_set_rate would not be called between enabling and
disabling continuous mode. What happened in this situation was that
timerfd_timer_enable_continuous saved off our previously set timer (in this
situation a 0 timer, meaning it never runs out). Then timerfd_timer_disable_continuous
would restore this 0 timer, even though it logically should set the timer to be whatever
was set in timerfd_timer_set_rate.
Now the behavior in timerfd_timer_set_rate is to overwrite the saved timer that may
or may not have been set in timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called, this will not harm the
operation.
Thanks to Terry Wilson for discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
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r168941 | twilson | 2009-01-16 23:16:23 +0100 (Fri, 16 Jan 2009) | 19 lines
Merged revisions 168716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines
Convert call to park_call_full to masq_park_call_announce
Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash.
(closes issue #14215)
Reported by: waverly360
Tested by: otherwiseguy
(closes issue #14228)
Reported by: kobaz
Tested by: otherwiseguy
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r168976 | mmichelson | 2009-01-16 23:43:09 +0100 (Fri, 16 Jan 2009) | 26 lines
Merged revisions 168975 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
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r169044 | twilson | 2009-01-17 01:03:39 +0100 (Sat, 17 Jan 2009) | 8 lines
Fix port :0 added to SIP INVITE URI when outboundproxy used
(closes issue #14233)
Reported by: chris-mac
Patches:
asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich, chris-mac, otherwiseguy
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r169080 | twilson | 2009-01-17 02:56:36 +0100 (Sat, 17 Jan 2009) | 8 lines
Fix qualify for TCP peer
(closes issue #14192)
Reported by: pabelanger
Patches:
asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich
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r169116 | seanbright | 2009-01-17 03:52:30 +0100 (Sat, 17 Jan 2009) | 8 lines
Change intializer types. Found while working on asterisk-cpp. I have a new
favorite error message from g++:
pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
initializers not supported
I like it when compilers are apologetic.
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r169153 | dbailey | 2009-01-17 19:26:44 +0100 (Sat, 17 Jan 2009) | 3 lines
Add discriminator for when ring pulse alert signal is used to preface MWI spills
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
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r169211 | mmichelson | 2009-01-19 16:54:06 +0100 (Mon, 19 Jan 2009) | 21 lines
Merged revisions 169210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines
Prevent a crash in chan_local due to a potential NULL pointer dereference
Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.
(closes issue #14189)
Reported by: sascha
Patches:
14189.patch uploaded by putnopvut (license 60)
Tested by: sascha
........
................
r169277 | dbailey | 2009-01-19 17:33:41 +0100 (Mon, 19 Jan 2009) | 9 lines
Add enhanced MWI generation to take advantage of new dahdi line reversal MWI ability.
(closes issue #14104)
Reported by: alecdavis
Patches:
asttrunk-14104.diff2.txt uploaded by dbailey (license )
chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dbailey
................
r169325 | dbailey | 2009-01-19 19:22:44 +0100 (Mon, 19 Jan 2009) | 3 lines
Get rid of magic number and replace with DAHDI_VMWI_NUMBER_MASK when
determining the number of messages pending for MWI call
................
r169327 | mvanbaak | 2009-01-19 19:36:24 +0100 (Mon, 19 Jan 2009) | 11 lines
Make asterisk compile on non-amd64 versions of OpenBSD.
The HW_PHYSMEM64 is only available in latest OpenBSD and/or amd64 versions of OpenBSD.
Use HW_PHYSMEM when HW_PHYSMEM64 is not available.
(closes issue #14129)
Reported by: ys
Patches:
2009011600_physmem64.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, jtodd
................
r169365 | tilghman | 2009-01-19 21:05:52 +0100 (Mon, 19 Jan 2009) | 11 lines
Merged revisions 169364 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines
Truncate userevents at the end of a line, when the command exceeds the buffer.
(closes issue #14278)
Reported by: fnordian
........
................
r169367 | mvanbaak | 2009-01-19 21:09:11 +0100 (Mon, 19 Jan 2009) | 15 lines
Redo the event-based MWI in chan_skinny.
Dan saw regular segfaults with the old implementation and
rewrote it to make it really eventbased.
I altered it to be trunk compatible and wedhorn gave some feedback
and ideas how to make it even better.
(closes issue #13821)
Reported by: DEA
Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3)
Tested by: mvanbaak, DEA
"no probs by me" from wedhorn
................
r169369 | mvanbaak | 2009-01-19 21:14:27 +0100 (Mon, 19 Jan 2009) | 7 lines
fix assignment in swapmode plug.
Spotted and fix provided by ys
(closes issue #14129)
Reported by: ys
Tested by: ys
................
r169438 | kpfleming | 2009-01-19 22:42:46 +0100 (Mon, 19 Jan 2009) | 5 lines
ast_str_SQLGetData is *not* part of the ast_str API, it's part of the ast_odbc API and just happens to use an ast_str as the buffer; move all of it to res_odbc.c and res_odbc.h, renaming appropriately
along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API
................
Added:
team/phsultan/jabberreceive/contrib/scripts/sip_nat_settings
- copied unchanged from r169438, trunk/contrib/scripts/sip_nat_settings
Modified:
team/phsultan/jabberreceive/ (props changed)
team/phsultan/jabberreceive/CHANGES
team/phsultan/jabberreceive/CREDITS
team/phsultan/jabberreceive/Makefile
team/phsultan/jabberreceive/apps/app_disa.c
team/phsultan/jabberreceive/apps/app_meetme.c
team/phsultan/jabberreceive/apps/app_minivm.c
team/phsultan/jabberreceive/apps/app_page.c
team/phsultan/jabberreceive/apps/app_queue.c
team/phsultan/jabberreceive/apps/app_read.c
team/phsultan/jabberreceive/apps/app_readexten.c
team/phsultan/jabberreceive/apps/app_userevent.c
team/phsultan/jabberreceive/apps/app_voicemail.c
team/phsultan/jabberreceive/build_tools/menuselect-deps.in
team/phsultan/jabberreceive/cdr/cdr_adaptive_odbc.c
team/phsultan/jabberreceive/cdr/cdr_odbc.c
team/phsultan/jabberreceive/channels/chan_agent.c
team/phsultan/jabberreceive/channels/chan_alsa.c
team/phsultan/jabberreceive/channels/chan_dahdi.c
team/phsultan/jabberreceive/channels/chan_local.c
team/phsultan/jabberreceive/channels/chan_misdn.c
team/phsultan/jabberreceive/channels/chan_sip.c
team/phsultan/jabberreceive/channels/chan_skinny.c
team/phsultan/jabberreceive/channels/chan_unistim.c
team/phsultan/jabberreceive/channels/misdn/isdn_lib.c
team/phsultan/jabberreceive/configs/chan_dahdi.conf.sample
team/phsultan/jabberreceive/configs/extconfig.conf.sample
team/phsultan/jabberreceive/configs/indications.conf.sample
team/phsultan/jabberreceive/configs/sip.conf.sample
team/phsultan/jabberreceive/configure
team/phsultan/jabberreceive/configure.ac
team/phsultan/jabberreceive/contrib/scripts/autosupport
team/phsultan/jabberreceive/funcs/func_channel.c
team/phsultan/jabberreceive/funcs/func_logic.c
team/phsultan/jabberreceive/funcs/func_odbc.c
team/phsultan/jabberreceive/include/asterisk/autoconfig.h.in
team/phsultan/jabberreceive/include/asterisk/channel.h
team/phsultan/jabberreceive/include/asterisk/indications.h
team/phsultan/jabberreceive/include/asterisk/options.h
team/phsultan/jabberreceive/include/asterisk/res_odbc.h
team/phsultan/jabberreceive/include/asterisk/say.h
team/phsultan/jabberreceive/include/asterisk/strings.h
team/phsultan/jabberreceive/include/asterisk/tcptls.h
team/phsultan/jabberreceive/main/app.c
team/phsultan/jabberreceive/main/ast_expr2.c
team/phsultan/jabberreceive/main/ast_expr2.h
team/phsultan/jabberreceive/main/ast_expr2.y
team/phsultan/jabberreceive/main/asterisk.c
team/phsultan/jabberreceive/main/channel.c
team/phsultan/jabberreceive/main/cli.c
team/phsultan/jabberreceive/main/features.c
team/phsultan/jabberreceive/main/http.c
team/phsultan/jabberreceive/main/indications.c
team/phsultan/jabberreceive/main/manager.c
team/phsultan/jabberreceive/main/pbx.c
team/phsultan/jabberreceive/main/say.c
team/phsultan/jabberreceive/main/srv.c
team/phsultan/jabberreceive/main/taskprocessor.c
team/phsultan/jabberreceive/main/tcptls.c
team/phsultan/jabberreceive/main/udptl.c
team/phsultan/jabberreceive/makeopts.in
team/phsultan/jabberreceive/pbx/pbx_dundi.c
team/phsultan/jabberreceive/res/ael/pval.c
team/phsultan/jabberreceive/res/res_agi.c
team/phsultan/jabberreceive/res/res_config_odbc.c
team/phsultan/jabberreceive/res/res_http_post.c
team/phsultan/jabberreceive/res/res_indications.c
team/phsultan/jabberreceive/res/res_odbc.c
team/phsultan/jabberreceive/res/res_timing_timerfd.c
team/phsultan/jabberreceive/res/snmp/agent.c
team/phsultan/jabberreceive/sounds/Makefile
Propchange: team/phsultan/jabberreceive/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/phsultan/jabberreceive/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/phsultan/jabberreceive/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jan 20 04:40:23 2009
@@ -1,1 +1,1 @@
-/trunk:1-168247
+/trunk:1-169481
Modified: team/phsultan/jabberreceive/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/CHANGES?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/CHANGES (original)
+++ team/phsultan/jabberreceive/CHANGES Tue Jan 20 04:40:23 2009
@@ -40,7 +40,12 @@
version received is different from the current SDP session version. This
option is required to interoperate with devices that have non-standard SDP
session version implementations (observed with Microsoft OCS). This option
- is diabled by default.
+ is disabled by default.
+ * The parsing of register => lines in sip.conf has been modified to allow a port
+ to be present in the "user" portion. Please see the sip.conf.sample file for more
+ information
+ * Added a function to remove SIP headers added in the dialplan before the
+ first INVITE is generated - SIPRemoveHeader()
Skinny Changes
--------------
@@ -116,6 +121,9 @@
timezones, especially if those daylight savings time ranges vary from your
machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
includes.
+ * The contrib/scripts/ directory now has a script called sip_nat_settings that will
+ give you the correct output for an asterisk box behind nat. It will give you the
+ externhost and localnet settings.
* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
XMPP text messages to the remote JID.
Modified: team/phsultan/jabberreceive/CREDITS
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/CREDITS?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/CREDITS (original)
+++ team/phsultan/jabberreceive/CREDITS Tue Jan 20 04:40:23 2009
@@ -192,6 +192,8 @@
- See http://voip-info.org/users/view/sergee
serg(AT)voipsolutions.ru
+Klaus Darillon - the SIPremoveHeader function in chan_sip
+
=== OTHER CONTRIBUTIONS ===
John Todd - Monkey sounds and associated teletorture prompt
Michael Jerris - bug marshaling
Modified: team/phsultan/jabberreceive/Makefile
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/Makefile?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/Makefile (original)
+++ team/phsultan/jabberreceive/Makefile Tue Jan 20 04:40:23 2009
@@ -726,6 +726,7 @@
echo ";rungroup = asterisk ; The group to run as" ; \
echo ";lightbackground = yes ; If your terminal is set for a light-colored background" ; \
echo "documentation_language = en_US ; Set the Language you want Documentation displayed in. Value is in the same format as locale names" ; \
+ echo ";hideconnect = yes ; Hide messages displayed when a remote console connects and disconnects" ; \
echo "" ; \
echo "; Changing the following lines may compromise your security." ; \
echo ";[files]" ; \
Modified: team/phsultan/jabberreceive/apps/app_disa.c
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/apps/app_disa.c?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/apps/app_disa.c (original)
+++ team/phsultan/jabberreceive/apps/app_disa.c Tue Jan 20 04:40:23 2009
@@ -124,7 +124,7 @@
static void play_dialtone(struct ast_channel *chan, char *mailbox)
{
- const struct ind_tone_zone_sound *ts = NULL;
+ const struct tone_zone_sound *ts = NULL;
if(ast_app_has_voicemail(mailbox, NULL))
ts = ast_get_indication_tone(chan->zone, "dialrecall");
else
Modified: team/phsultan/jabberreceive/apps/app_meetme.c
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/apps/app_meetme.c?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/apps/app_meetme.c (original)
+++ team/phsultan/jabberreceive/apps/app_meetme.c Tue Jan 20 04:40:23 2009
@@ -2187,10 +2187,12 @@
user->user_no = AST_LIST_LAST(&conf->userlist)->user_no + 1;
if (rt_schedule && conf->maxusers)
- if (user->user_no > conf->maxusers) {
+ if (conf->users >= conf->maxusers) {
/* Sorry, but this confernce has reached the participant limit! */
if (!ast_streamfile(chan, "conf-full", chan->language))
ast_waitstream(chan, "");
+ ast_mutex_unlock(&conf->playlock);
+ user->user_no = 0;
goto outrun;
}
Modified: team/phsultan/jabberreceive/apps/app_minivm.c
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/apps/app_minivm.c?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/apps/app_minivm.c (original)
+++ team/phsultan/jabberreceive/apps/app_minivm.c Tue Jan 20 04:40:23 2009
@@ -2261,7 +2261,7 @@
if(!(vmu = find_account(domain, username, TRUE))) {
/* We could not find user, let's exit */
ast_log(LOG_WARNING, "Could not allocate temporary memory for '%s@%s'\n", username, domain);
- pbx_builtin_setvar_helper(chan, "MVM_NOTIFY_STATUS", "FAILED");
+ pbx_builtin_setvar_helper(chan, "MVM_ACCMESS_STATUS", "FAILED");
return -1;
}
@@ -2292,7 +2292,7 @@
if(ast_test_flag(vmu, MVM_ALLOCED))
free_user(vmu);
- pbx_builtin_setvar_helper(chan, "MVM_NOTIFY_STATUS", "SUCCESS");
+ pbx_builtin_setvar_helper(chan, "MVM_ACCMESS_STATUS", "SUCCESS");
/* Ok, we're ready to rock and roll. Return to dialplan */
return 0;
Modified: team/phsultan/jabberreceive/apps/app_page.c
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/apps/app_page.c?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/apps/app_page.c (original)
+++ team/phsultan/jabberreceive/apps/app_page.c Tue Jan 20 04:40:23 2009
@@ -117,7 +117,6 @@
AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
});
-#define MAX_DIALS 128
static int page_exec(struct ast_channel *chan, void *data)
{
@@ -127,7 +126,8 @@
unsigned int confid = ast_random();
struct ast_app *app;
int res = 0, pos = 0, i = 0;
- struct ast_dial *dials[MAX_DIALS];
+ struct ast_dial **dial_list;
+ unsigned int num_dials;
int timeout = 0;
char *parse;
@@ -166,6 +166,18 @@
snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
+
+ /* Count number of extensions in list by number of ampersands + 1 */
+ num_dials = 1;
+ tmp = args.devices;
+ while (*tmp && *tmp++ == '&') {
+ num_dials++;
+ }
+
+ if (!(dial_list = ast_calloc(num_dials, sizeof(void *)))) {
+ ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(void *) * num_dials));
+ return -1;
+ }
/* Go through parsing/calling each device */
while ((tech = strsep(&args.devices, "&"))) {
@@ -222,7 +234,7 @@
ast_dial_run(dial, chan, 1);
/* Put in our dialing array */
- dials[pos++] = dial;
+ dial_list[pos++] = dial;
}
if (!ast_test_flag(&flags, PAGE_QUIET)) {
@@ -239,7 +251,7 @@
/* Go through each dial attempt cancelling, joining, and destroying */
for (i = 0; i < pos; i++) {
- struct ast_dial *dial = dials[i];
+ struct ast_dial *dial = dial_list[i];
/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
ast_dial_join(dial);
Modified: team/phsultan/jabberreceive/apps/app_queue.c
URL: http://svn.digium.com/svn-view/asterisk/team/phsultan/jabberreceive/apps/app_queue.c?view=diff&rev=169482&r1=169481&r2=169482
==============================================================================
--- team/phsultan/jabberreceive/apps/app_queue.c (original)
[... 7632 lines stripped ...]
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