[svn-commits] rmudgett: branch group/issue14068 r209738 - in /team/group/issue14068: ./ add...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jul 31 17:25:02 CDT 2009
Author: rmudgett
Date: Fri Jul 31 17:24:34 2009
New Revision: 209738
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=209738
Log:
Merged revisions 203338,203376,203381,203402,203443-203444,203479,203508,203525,203534,203569,203605,203638,203640,203672,203699,203702,203710,203720-203721,203735,203779,203783,203802,203842,203846,203853,203909,203960,203962,203985,204013,204069,204118-204119,204143,204171,204217,204247,204301,204355,204413,204415,204417-204420,204422-204423,204428,204440,204470,204475,204530,204532,204561,204563,204622,204654,204710,204749,204807,204835,204893,204919,204948,204986,205014,205047,205086,205118,205120,205150-205151,205196,205214,205216,205221,205254,205291,205350,205410,205412,205469,205479,205532,205562,205600,205666,205696,205700,205770,205776,205780,205840,205878,205939,205985,206021,206049,206092,206094,206127,206185,206225,206280,206341,206386,206455,206489-206490,206566-206567,206603,206636,206702,206707,206767-206768,206808,206868,206873,206877,206939,206998,207029,207034,207093,207095,207156,207224-207225,207255,207285,207317-207318,207361,207424,207484,207522,207551,207599,207680,207723,207854,207902,207925,207934,207946,207950,208017-208018,208052,208113,208151,208155,208193,208229,208263,208267,208314,208383,208388,208464,208504,208542,208548,208588,208593,208630,208693,208706,208709,208749,208813,208848,208886,208924,208991,209056,209098,209132,209197,209235,209256,209279,209317,209331,209400,209453,209484,209516,209554,209619,209623,209673-209674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r203338 | twilson | 2009-06-25 15:25:39 -0500 (Thu, 25 Jun 2009) | 9 lines
Merged revisions 203311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines
Don't try to free NULL
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r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines
Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
Fix a case where CDR answer time could be before the start time involving parking.
(closes issue #13794)
Reported by: davidw
Patches:
13794.patch uploaded by murf (license 17)
13794.patch.160 uploaded by murf (license 17)
Tested by: murf, dbrooks
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r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines
Merged revisions 203380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines
I didn't see that Mark already fixed the underlying issue!
Yay for removing useless code.
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r203402 | jpeeler | 2009-06-25 16:22:12 -0500 (Thu, 25 Jun 2009) | 2 lines
Remove some unnecessary code and update sample config file with respect to GR-303.
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r203443 | rmudgett | 2009-06-25 16:34:18 -0500 (Thu, 25 Jun 2009) | 1 line
Picking nits
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r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines
fixes a few redundant conditions
(issue #15269)
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r203479 | jpeeler | 2009-06-25 17:48:33 -0500 (Thu, 25 Jun 2009) | 1 line
make sure chan_dahdi compiles with only libss7 and not libpri installed
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r203508 | seanbright | 2009-06-25 18:54:03 -0500 (Thu, 25 Jun 2009) | 5 lines
Move syslog utility functions into a separate file so they can be re-used.
This has the pleasant side effect of cleaning up the header inclusion process
in logger.c.
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r203525 | russell | 2009-06-25 19:21:09 -0500 (Thu, 25 Jun 2009) | 2 lines
Convert spaces to tabs for indentation.
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r203534 | russell | 2009-06-25 19:23:55 -0500 (Thu, 25 Jun 2009) | 2 lines
One more formatting nit ... use spaces for inline indentation.
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r203569 | seanbright | 2009-06-25 22:06:06 -0500 (Thu, 25 Jun 2009) | 2 lines
Add checks in configure for non-POSIX syslog facilities.
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r203605 | seanbright | 2009-06-26 08:00:35 -0500 (Fri, 26 Jun 2009) | 5 lines
Add functions to map syslog facilities and priorities constants to strings.
Also change the default casing of the string contants to lowercase. This really
just saves us from have to lowercase them later when displaying them.
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r203638 | russell | 2009-06-26 10:28:53 -0500 (Fri, 26 Jun 2009) | 14 lines
Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
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r203640 | russell | 2009-06-26 10:42:26 -0500 (Fri, 26 Jun 2009) | 2 lines
Note a new API call, and one that changed in doxygen.
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r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines
Check if polarityonanswerdelay has elapsed before setting a channel as answered
after a polarity reversal.
Previously on a polarity switch event chan_dahdi would set the channel
immediately as answered. This would cause problems if a polarity reversal
occurred when the line was picked up as the dial would not have yet occurred.
Now if the polarity reversal occurs before delay has elapsed after coming off
hook or an answer, it is ignored. Also, some refactoring was done in
_handle_event.
(closes issue #13917)
Reported by: alecdavis
Patches:
chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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r203699 | file | 2009-06-26 14:27:24 -0500 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between application and channel.
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r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
Make invalid hints report Unavailable instead of Idle.
(closes issue #14413)
Reported by: pj
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r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines
moving debug message from level 0 to 1.
(closes issue #15404)
Reported by: leobrown
Patches:
iax_codec_debug.patch uploaded by leobrown (license 541)
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r203720 | dbrooks | 2009-06-26 15:11:47 -0500 (Fri, 26 Jun 2009) | 22 lines
Blocked revisions 203719 via svnmerge
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r203719 | dbrooks | 2009-06-26 15:03:42 -0500 (Fri, 26 Jun 2009) | 16 lines
Fixing voicemail's error in checking max silence vs min message length
Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.
Also, the inequality was reversed. The warning, if triggered, was "Max silence should
be less than minmessage or you may get empty messages", which should have been logged
if max silence was greater than minmessage, but the check was for less than.
Also, conforming if statement to coding guidelines.
closes issue #15331)
Reported by: markd
Review: https://reviewboard.asterisk.org/r/293/
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r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines
Fixing voicemail's error in checking max silence vs min message length
Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.
Also, the inequality was reversed. The warning, if triggered, was "Max silence should
be less than minmessage or you may get empty messages", which should have been logged
if max silence was greater than minmessage, but the check was for less than.
Also, conforming if statement to coding guidelines.
closes issue #15331)
Reported by: markd
Review: https://reviewboard.asterisk.org/r/293/
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r203735 | file | 2009-06-26 15:19:49 -0500 (Fri, 26 Jun 2009) | 6 lines
Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control.
(closes issue #8855)
Reported by: mikma
Tested by: klaus3000, file
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r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
Ensure the TCP read buffer is fully initialized before handling each packet.
(closes issue #14452)
Reported by: umberto71
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r203783 | mmichelson | 2009-06-26 15:52:19 -0500 (Fri, 26 Jun 2009) | 8 lines
Add timestamp to response to "Ping" manager action.
(closes issue #14596)
Reported by: JimDickenson
Patches:
pong2.diff uploaded by JimDickenson (license 710)
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r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines
Merged revisions 203785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
Don't fast forward past the end of a message.
This is nice change for users of the voicemail application. If someone gets a
little carried away with fast forwarding through a message, they can easily
get to the end and accidentally exit the voicemail application by hitting the
fast forward key during the following prompt.
This adds some safety by not allowing a fast forward past the end of a message.
(closes issue #14554)
Reported by: lacoursj
Patches:
21761.patch uploaded by lacoursj (license 707)
Tested by: lacoursj
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r203842 | russell | 2009-06-26 16:48:41 -0500 (Fri, 26 Jun 2009) | 7 lines
Add 's' option to ChanSpy, which makes the app exit when no channels are left to spy on.
(closes issue #14594)
Reported by: JimDickenson
Patches:
chanspy.diff uploaded by JimDickenson (license 710)
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r203846 | seanbright | 2009-06-26 17:08:05 -0500 (Fri, 26 Jun 2009) | 14 lines
Add a new module, cdr_syslog, which allows writing CDRs to syslog.
The original patch for this was written by Brett Bryant, and I split it out into
it's own module.
(closes issue #12876)
Reported by: bbryant
Patches:
06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
05212009_cdr_syslog.patch uploaded by seanbright (license 71)
Tested by: seanbright
Review: https://reviewboard.asterisk.org/r/297/
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r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines
Merged revisions 203848 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines
Make sure to recreate the dahdi pseudo channel after dahdi restart
(closes issue #14477)
Reported by: timking
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r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines
Merged revisions 203908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines
The ISDN CPE side should not exclusively pick B channels normally.
Before this patch, Asterisk unconditionally picked B channels exclusively
on the CPE side and normally allowed alternative B channels on the network
side. Now Asterisk does the opposite.
Reasons for the CPE side to normally not pick B channels exclusively:
* For CPE point-to-multipoint mode (i.e. phone side), the CPE side does
not have enough information to exclusively pick B channels. (There may be
other devices on the line.)
* Q.931 gives preference to the network side picking B channels.
* Some telcos require the CPE side to not pick B channels exclusively.
(closes issue #14383)
Reported by: mbrancaleoni
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r203960 | russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
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r203962 | russell | 2009-06-27 05:04:51 -0500 (Sat, 27 Jun 2009) | 8 lines
Only update total silence counter after a counter reset.
(closes issue #2264)
Reported by: pfn
Patches:
silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810)
Tested by: pfn
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r203985 | seanbright | 2009-06-27 15:26:01 -0500 (Sat, 27 Jun 2009) | 1 line
Another CHANGES spelling fix.
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r204013 | mmichelson | 2009-06-29 10:04:39 -0500 (Mon, 29 Jun 2009) | 11 lines
Blocked revisions 204012 via svnmerge
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r204012 | mmichelson | 2009-06-29 10:04:17 -0500 (Mon, 29 Jun 2009) | 6 lines
Place unlock of mutex in an else block so that it does not get unlocked twice.
(closes issue #15400)
Reported by: aragon
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r204069 | tilghman | 2009-06-29 12:15:15 -0500 (Mon, 29 Jun 2009) | 7 lines
Remove invalid entries in the config.
This might seem like a legitimate comment that merely needed semicolon
prefixes, but in reality, the adaptive layer is designed to allow arbitrary
CDR variables, without needing the use of a userfield to store multiple items.
It's therefore not only invalid syntax but also goes against the intent of the
adaptive method.
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r204118 | tilghman | 2009-06-29 12:56:29 -0500 (Mon, 29 Jun 2009) | 2 lines
Allow trunk to once again compile under MALLOC_DEBUG
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r204119 | seanbright | 2009-06-29 13:05:27 -0500 (Mon, 29 Jun 2009) | 1 line
Add common headers to CEL related configs.
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r204143 | seanbright | 2009-06-29 13:44:44 -0500 (Mon, 29 Jun 2009) | 7 lines
Get app_rpt compiling again. I doubt seriously that it actually works.
Also, the code in this module is horrendous and we should remove it from the
tree. I'm not sure who is supposed to be maintaning this thing, but they
clearly are not. I don't see the sense of leaving it in the main tree. If it
lives *anywhere* it should be in addons.
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r204171 | tilghman | 2009-06-29 14:36:57 -0500 (Mon, 29 Jun 2009) | 9 lines
Blocked revisions 204170 via svnmerge
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r204170 | tilghman | 2009-06-29 14:36:01 -0500 (Mon, 29 Jun 2009) | 3 lines
Revision 189537 was supposed to make 1.4 more correct. Instead, it broke func_odbc. Reverting.
(closes issue #15317, issue #14614)
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r204217 | seanbright | 2009-06-29 15:29:10 -0500 (Mon, 29 Jun 2009) | 1 line
Reorganize this adaptive CEL config a bit.
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r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 34 lines
Merged revisions 204243,204246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
Fix a problem where chan_sip would ignore "old" but valid responses.
chan_sip has had a problem for quite a long time that would manifest when
Asterisk would send multiple SIP responses on the same dialog before receiving
a response. The problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two requests out,
and a response arrived for the first request sent, then Asterisk would ignore
the response. The result was that Asterisk would continue retransmitting the
requests and ignoring the responses until the maximum number of retransmissions
had been reached.
The fix here is to rearrange the code a bit so that instead of simply comparing
the sequence number of the response to our latest outgoing sequence number, we
walk our list of outstanding packets and determine if there is a match. If there is,
we continue. If not, then we ignore the response.
In doing this, I found a few completely useless variables that I have now removed.
(closes issue #11231)
Reported by: flefoll
Review: https://reviewboard.asterisk.org/r/298
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r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
Fix build oops.
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r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines
Merged revisions 204300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
Add error message so that it is clear why a SIP peer was not processed when
a DNS lookup fails on a host or outboundproxy.
(closes issue #13432)
Reported by: p_lindheimer
Patches:
outboundproxy.patch uploaded by p (license 558)
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r204355 | seanbright | 2009-06-29 18:50:46 -0500 (Mon, 29 Jun 2009) | 2 lines
A few const changes in app_meetme.c that I noticed while browsing the source.
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r204413 | russell | 2009-06-30 11:40:38 -0500 (Tue, 30 Jun 2009) | 12 lines
Move Asterisk-addons modules into the main Asterisk source tree.
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
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r204415 | kpfleming | 2009-06-30 12:04:35 -0500 (Tue, 30 Jun 2009) | 8 lines
Add-ons related build system improvements.
Ensure that add-on modules can be embedded, fix up Makefile.moddir_rules
to allow module directory Makefiles to more easily specify the modules to
be built, and explicitly list the addons modules in its Makefile, since
the module names don't follow any pattern.
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r204417 | russell | 2009-06-30 12:08:14 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename app_addon_sql_mysql to app_mysql
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r204418 | russell | 2009-06-30 12:09:04 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename cdr_addon_mysql to cdr_mysql
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r204419 | russell | 2009-06-30 12:10:45 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename mysql.conf to app_mysql.conf, make module support both names
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r204420 | russell | 2009-06-30 12:11:31 -0500 (Tue, 30 Jun 2009) | 2 lines
Make addons build last - this is for Qwell.
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r204422 | russell | 2009-06-30 12:15:09 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename res_mysql.conf to res_config_mysql.conf, make module support both
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r204423 | russell | 2009-06-30 12:16:56 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename mobile.conf to chan_mobile.conf, make module support old name, too
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r204428 | russell | 2009-06-30 12:18:18 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename ooh323.conf to chan_ooh323.conf, make module support both names
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r204440 | russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing .sample).
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r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines
Recorded merge of revisions 204469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346)
Reported by: volivier
Patches:
20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: volivier
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r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines
Merged revisions 204474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line
Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing.
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r204530 | mmichelson | 2009-06-30 14:55:59 -0500 (Tue, 30 Jun 2009) | 10 lines
Remove some bogus deadlock avoidance code from local_attended_transfer.
First of all, the code was unnecessary. The goal was to lock a channel
which was already locked. Second, the assumption of the deadlock avoidance
loop was that the sip_pvt was already locked and we were trying to get the
channel lock. The problem is that the sip_pvt was unlocked a few lines above.
Basically, I'm removing 5 lines of no-op.
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r204532 | mmichelson | 2009-06-30 14:59:20 -0500 (Tue, 30 Jun 2009) | 5 lines
Move the masquerade in local_attended_transfer to a point where we hold the channel lock.
Masquerading without the channel's lock held is a *horrible* idea.
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r204561 | seanbright | 2009-06-30 15:39:39 -0500 (Tue, 30 Jun 2009) | 1 line
Remove an unnecessary #ifdef
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r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines
Merged revisions 204556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022)
Reported by: greenfieldtech
Patches:
20090519__issue15022.diff.txt uploaded by tilghman (license 14)
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r204622 | seanbright | 2009-07-01 11:06:18 -0500 (Wed, 01 Jul 2009) | 2 lines
A bunch of CODING_GUIDELINES related fixes. Not even close to done.
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r204654 | rbrindley | 2009-07-01 14:47:38 -0500 (Wed, 01 Jul 2009) | 4 lines
- cfgbasic.html has been replaced by index.html in the GUI for some time now
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r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
Merged revisions 204681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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r204749 | seanbright | 2009-07-02 12:46:14 -0500 (Thu, 02 Jul 2009) | 21 lines
Support setting and receiving Reverse Charging Indication over ISDN PRI.
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
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r204807 | mnicholson | 2009-07-02 15:37:16 -0500 (Thu, 02 Jul 2009) | 2 lines
Moved trigger for BRIDGE_END CEL event so that it is more accurate.
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r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines
Merged revisions 204834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
Removed confusing warning message "Got Busy in Connected State"
If an incoming mISDN call is answered with the Answer application and a
subsequent Dial gets a busy endpoint then it is valid for that already
connected channel to get the busy indication. Asterisk will play the busy
tones until the dialplan plays something else or hangs up the call.
(closes issue #11974)
Reported by: fvdb
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r204893 | seanbright | 2009-07-02 21:02:50 -0500 (Thu, 02 Jul 2009) | 1 line
Wrap rtp_engine.h header comments to 80 characters.
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r204919 | seanbright | 2009-07-03 10:44:01 -0500 (Fri, 03 Jul 2009) | 5 lines
Add a configure check for Reverse Charging Indication support in LibPRI.
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
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r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines
Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels.
This change allows applications that request T.38 negotiation on a channel that
does not support it to get the proper indication that it is not supported, rather
than thinking that negotiation was started when it was not.
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r204986 | tilghman | 2009-07-06 16:37:39 -0500 (Mon, 06 Jul 2009) | 14 lines
Merged revisions 981 via svnmerge from
https://origsvn.digium.com/svn/asterisk-addons/branches/1.4
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r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul 2009) | 7 lines
Don't reset reconnect time, unless a reconnect really occurred.
(closes issue #15375)
Reported by: kowalma
Patches:
20090628__issue15375.diff.txt uploaded by tilghman (license 14)
Tested by: kowalma, jacco
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r205014 | mnicholson | 2009-07-06 18:24:57 -0500 (Mon, 06 Jul 2009) | 2 lines
Add CEL transfer events to analog (chan_dahdi) transfers.
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r205047 | mnicholson | 2009-07-07 13:24:13 -0500 (Tue, 07 Jul 2009) | 2 lines
Fix a deadlock in sig_analog
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r205086 | tilghman | 2009-07-07 16:10:14 -0500 (Tue, 07 Jul 2009) | 4 lines
Permit setting custom headers from the peer definition.
(closes issue #14059)
Reported by: fnordian
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r205118 | rizzo | 2009-07-08 09:45:15 -0500 (Wed, 08 Jul 2009) | 3 lines
FreeBSD now has autoconf 2.62 in the ports, 2.61 has disappeared.
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r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
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r205150 | russell | 2009-07-08 10:54:42 -0500 (Wed, 08 Jul 2009) | 14 lines
Blocked revisions 205149 via svnmerge
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r205149 | russell | 2009-07-08 10:54:21 -0500 (Wed, 08 Jul 2009) | 8 lines
Make OpenSSL usage thread-safe.
OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
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r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
Use tabs instead of spaces for indentation.
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r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines
Merged revisions 205188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines
Add redirection warnings for the invalid language codes previously removed.
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r205214 | seanbright | 2009-07-08 11:43:12 -0500 (Wed, 08 Jul 2009) | 2 lines
Fix a few compilation problems found when building Asterisk against uClibc.
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r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines
Merged revisions 205215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines
ast_samp2tv needs floating point for 16khz audio
In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
The .5 is currently stripped off because we don't calculate
using floating points. This causes madness with 16khz audio.
(issue ABE-1899)
Review: https://reviewboard.asterisk.org/r/305/
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r205221 | tilghman | 2009-07-08 11:59:32 -0500 (Wed, 08 Jul 2009) | 2 lines
Oops, fixing build
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r205254 | dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
Fixes Park() argument handling
Park() was not respecting the arguments passed to it. Any extension/context/priority
given to it was being ignored. This patch remedies this.
(closes issue #15380)
Reported by: DLNoah
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r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines
Merged revisions 205288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line
Update config.guess and config.sub from the savannah.gnu.org git repo.
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r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines
Merged revisions 205349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
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r205410 | dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
missing comma in devstatestring array
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r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines
Merged revisions 205409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change
fixes a compile time error with chan_vpb as well.
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r205469 | mnicholson | 2009-07-08 18:07:09 -0500 (Wed, 08 Jul 2009) | 5 lines
Fix a CEL related regression with hints updating by subscribing to AST_DEVICE_STATE instead of AST_DEVICE_STATE_CHANGED.
(closes issue #15440)
Reported by: lmsteffan
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r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines
Merged revisions 205471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This
is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas
that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/
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r205532 | mvanbaak | 2009-07-09 03:31:24 -0500 (Thu, 09 Jul 2009) | 5 lines
pthread_self returns a pthread_t which is not an unsigned int on all
pthread implementations. Casting it to an unsigned int fixes compiler warnings.
Tested on OpenBSD and Linux both 32 and 64 bit
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r205562 | mvanbaak | 2009-07-09 09:10:01 -0500 (Thu, 09 Jul 2009) | 2 lines
make this compile again under devmode
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r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines
Merged revisions 205599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines
Changing ast_samp2tv to not use floating point.
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r205666 | mnicholson | 2009-07-09 15:04:43 -0500 (Thu, 09 Jul 2009) | 4 lines
Convert func_odbc to use ast_dummy_alloc_channel()
Review: https://reviewboard.asterisk.org/r/290/
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
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r205700 | mnicholson | 2009-07-09 16:32:31 -0500 (Thu, 09 Jul 2009) | 5 lines
Fix mbl_fixup() in chan_mobile to update newchan->tech_pvt instead of oldchan.
(closes issue #15299)
Reported by: nikkk
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r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines
Fix some remaining T.38 negotiation problems in app_fax.
Revision 205696 did not quite fix all the issues with the T.38 negotiation
changes and app_fax; this patch corrects them, along with a couple of other
minor issues.
(closes issue #15480)
Reported by: dimas
Patches:
test2-15480.patch uploaded by dimas (license 88)
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r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
Merged revisions 205775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
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r205780 | kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11 lines
Eliminate extraneous LOG_DEBUG messages generated by app_fax.
The transmit_audio() and transmit_t38() functions in app_fax have processing
loops that are supposed to wait for frames to arrive on the channel and then
handle them, but they also have short timeouts so that the loops can have
watchdog timers and do other required processing. This commit changes the loops
to not actually call ast_read() and attempt to process the returned frame
unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages
and slightly improving performance.
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r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines
Merged revisions 205804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines
SIP registration auth loop caused by stale nonce
If an endpoint sends two registration requests in a very short
period of time with the same nonce, both receive 401 responses
from Asterisk, each with a different nonce (the second 401
containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match
the current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if
the endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a
newly generated nonce... This loop goes on and on.
There appears to be a simple fix for this. If the nonce from
the request does not match our nonce, but is a good response
to a previous nonce, instead of sending a 401 with a newly
generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no
nonce can be responded to twice though.
(closes issue #15102)
Reported by: Jamuel
Patches:
patch-bug_0015102 uploaded by Jamuel (license 809)
nonce_sip.diff uploaded by dvossel (license 671)
Tested by: Jamuel
Review: https://reviewboard.asterisk.org/r/289/
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r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines
Merged revisions 205877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines
Merged revisions 205776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
Merged revisions 205775 via svnmerge from
[... 24922 lines stripped ...]
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