[svn-commits] junky: branch junky/cli-tls r204382 - in /team/junky/cli-tls: ./ apps/ bridge...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jun 29 23:29:07 CDT 2009
Author: junky
Date: Mon Jun 29 23:28:56 2009
New Revision: 204382
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=204382
Log:
Merged revisions 203116,203126,203188,203227,203231,203258,203304,203338,203376,203381,203402,203443-203444,203479,203508,203525,203534,203569,203605,203638,203640,203672,203699,203702,203710,203720-203721,203735,203779,203783,203802,203842,203846,203853,203909,203960,203962,203985,204013,204069,204118-204119,204143,204171,204217,204247,204301,204355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r203116 | russell | 2009-06-25 12:04:10 -0400 (Thu, 25 Jun 2009) | 18 lines
Merged revisions 203115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
Resolve a crash related to a T.38 reinvite race condition.
This change resolves a crash observed locally during some T.38 testing.
A call was set up using a call file, and when the T.38 reinvite came in,
the channel state was still AST_STATE_DOWN. The reason is explained by
a comment in the code that previously lived in the handling of
AST_STATE_RINGING. This change modifies the logic to handle the same
race condition for any channel state that is not UP.
(closes ABE-1895)
........
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r203126 | dbailey | 2009-06-25 12:18:48 -0400 (Thu, 25 Jun 2009) | 12 lines
Insure ring cadence is set for fxs ports
Moved SETCADENCE ioctl call to before call into new analog signal module
to insure that it gets set.
(closes issue #15381)
Reported by: alecdavis
Patches:
fix15381.diff uploaded by dbailey (license 819)
Tested by: dbailey
................
r203188 | seanbright | 2009-06-25 13:01:32 -0400 (Thu, 25 Jun 2009) | 2 lines
Pass a logmsg to ast_log_vsyslog instead of separate arguments.
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r203227 | file | 2009-06-25 14:25:24 -0400 (Thu, 25 Jun 2009) | 7 lines
Add support for multicast RTP paging.
(closes issue #11797)
Reported by: macbrody
Review: https://reviewboard.asterisk.org/r/270/
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r203231 | mmichelson | 2009-06-25 14:53:12 -0400 (Thu, 25 Jun 2009) | 8 lines
Blocked revisions 203230 via svnmerge
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r203230 | mmichelson | 2009-06-25 13:52:22 -0500 (Thu, 25 Jun 2009) | 3 lines
Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.
........
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r203258 | qwell | 2009-06-25 15:22:46 -0400 (Thu, 25 Jun 2009) | 10 lines
Unmute when we get a dtmfup (we muted on dtmfdown) event.
This would occasionally cause one-way audio when using hardware DTMF detection.
(closes issue #14761)
Reported by: tzafrir
Patches:
v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas
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r203304 | jpeeler | 2009-06-25 15:54:12 -0400 (Thu, 25 Jun 2009) | 6 lines
New signaling module to handle PRI/BRI operations in chan_dahdi
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.
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r203338 | twilson | 2009-06-25 16:25:39 -0400 (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 17:04:55 -0400 (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 17:15:11 -0400 (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 17:22:12 -0400 (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 17:34:18 -0400 (Thu, 25 Jun 2009) | 1 line
Picking nits
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r203444 | dvossel | 2009-06-25 17:45:32 -0400 (Thu, 25 Jun 2009) | 4 lines
fixes a few redundant conditions
(issue #15269)
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r203479 | jpeeler | 2009-06-25 18:48:33 -0400 (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 19:54:03 -0400 (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 20:21:09 -0400 (Thu, 25 Jun 2009) | 2 lines
Convert spaces to tabs for indentation.
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r203534 | russell | 2009-06-25 20:23:55 -0400 (Thu, 25 Jun 2009) | 2 lines
One more formatting nit ... use spaces for inline indentation.
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r203569 | seanbright | 2009-06-25 23:06:06 -0400 (Thu, 25 Jun 2009) | 2 lines
Add checks in configure for non-POSIX syslog facilities.
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r203605 | seanbright | 2009-06-26 09:00:35 -0400 (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 11:28:53 -0400 (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 11:42:26 -0400 (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 15:03:25 -0400 (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 15:27:24 -0400 (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 15:31:14 -0400 (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 15:47:11 -0400 (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 16:11:47 -0400 (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 16:13:51 -0400 (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 16:19:49 -0400 (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 16:45:00 -0400 (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 16:52:19 -0400 (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 17:21:48 -0400 (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 17:48:41 -0400 (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 18:08:05 -0400 (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 18:11:31 -0400 (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 21:07:52 -0400 (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 05:51:45 -0400 (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 06:04:51 -0400 (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 16:26:01 -0400 (Sat, 27 Jun 2009) | 1 line
Another CHANGES spelling fix.
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r204013 | mmichelson | 2009-06-29 11:04:39 -0400 (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 13:15:15 -0400 (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 13:56:29 -0400 (Mon, 29 Jun 2009) | 2 lines
Allow trunk to once again compile under MALLOC_DEBUG
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r204119 | seanbright | 2009-06-29 14:05:27 -0400 (Mon, 29 Jun 2009) | 1 line
Add common headers to CEL related configs.
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r204143 | seanbright | 2009-06-29 14:44:44 -0400 (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 15:36:57 -0400 (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 16:29:10 -0400 (Mon, 29 Jun 2009) | 1 line
Reorganize this adaptive CEL config a bit.
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r204247 | mmichelson | 2009-06-29 17:48:54 -0400 (Mon, 29 Jun 2009) | 32 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
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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 18:50:35 -0400 (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 19:50:46 -0400 (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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Added:
team/junky/cli-tls/apps/app_celgenuserevent.c
- copied unchanged from r204355, trunk/apps/app_celgenuserevent.c
team/junky/cli-tls/cdr/cdr_syslog.c
- copied unchanged from r204355, trunk/cdr/cdr_syslog.c
team/junky/cli-tls/cel/ (props changed)
- copied from r204355, trunk/cel/
team/junky/cli-tls/cel/Makefile
- copied unchanged from r204355, trunk/cel/Makefile
team/junky/cli-tls/cel/cel_adaptive_odbc.c
- copied unchanged from r204355, trunk/cel/cel_adaptive_odbc.c
team/junky/cli-tls/cel/cel_custom.c
- copied unchanged from r204355, trunk/cel/cel_custom.c
team/junky/cli-tls/cel/cel_manager.c
- copied unchanged from r204355, trunk/cel/cel_manager.c
team/junky/cli-tls/cel/cel_pgsql.c
- copied unchanged from r204355, trunk/cel/cel_pgsql.c
team/junky/cli-tls/cel/cel_radius.c
- copied unchanged from r204355, trunk/cel/cel_radius.c
team/junky/cli-tls/cel/cel_sqlite3_custom.c
- copied unchanged from r204355, trunk/cel/cel_sqlite3_custom.c
team/junky/cli-tls/cel/cel_tds.c
- copied unchanged from r204355, trunk/cel/cel_tds.c
team/junky/cli-tls/channels/chan_multicast_rtp.c
- copied unchanged from r204355, trunk/channels/chan_multicast_rtp.c
team/junky/cli-tls/channels/sig_pri.c
- copied unchanged from r204355, trunk/channels/sig_pri.c
team/junky/cli-tls/channels/sig_pri.h
- copied unchanged from r204355, trunk/channels/sig_pri.h
team/junky/cli-tls/configs/cdr_syslog.conf.sample
- copied unchanged from r204355, trunk/configs/cdr_syslog.conf.sample
team/junky/cli-tls/configs/cel.conf.sample
- copied unchanged from r204355, trunk/configs/cel.conf.sample
team/junky/cli-tls/configs/cel_adaptive_odbc.conf.sample
- copied unchanged from r204355, trunk/configs/cel_adaptive_odbc.conf.sample
team/junky/cli-tls/configs/cel_custom.conf.sample
- copied unchanged from r204355, trunk/configs/cel_custom.conf.sample
team/junky/cli-tls/configs/cel_pgsql.conf.sample
- copied unchanged from r204355, trunk/configs/cel_pgsql.conf.sample
team/junky/cli-tls/configs/cel_sqlite3_custom.conf.sample
- copied unchanged from r204355, trunk/configs/cel_sqlite3_custom.conf.sample
team/junky/cli-tls/configs/cel_tds.conf.sample
- copied unchanged from r204355, trunk/configs/cel_tds.conf.sample
team/junky/cli-tls/doc/tex/cel-doc.tex
- copied unchanged from r204355, trunk/doc/tex/cel-doc.tex
team/junky/cli-tls/doc/tex/celdriver.tex
- copied unchanged from r204355, trunk/doc/tex/celdriver.tex
team/junky/cli-tls/include/asterisk/cel.h
- copied unchanged from r204355, trunk/include/asterisk/cel.h
team/junky/cli-tls/include/asterisk/syslog.h
- copied unchanged from r204355, trunk/include/asterisk/syslog.h
team/junky/cli-tls/main/cel.c
- copied unchanged from r204355, trunk/main/cel.c
team/junky/cli-tls/main/syslog.c
- copied unchanged from r204355, trunk/main/syslog.c
team/junky/cli-tls/res/res_rtp_multicast.c
- copied unchanged from r204355, trunk/res/res_rtp_multicast.c
Modified:
team/junky/cli-tls/ (props changed)
team/junky/cli-tls/CHANGES
team/junky/cli-tls/Makefile
team/junky/cli-tls/UPGRADE.txt
team/junky/cli-tls/apps/app_chanisavail.c
team/junky/cli-tls/apps/app_chanspy.c
team/junky/cli-tls/apps/app_confbridge.c
team/junky/cli-tls/apps/app_dial.c
team/junky/cli-tls/apps/app_directed_pickup.c
team/junky/cli-tls/apps/app_fax.c
team/junky/cli-tls/apps/app_followme.c
team/junky/cli-tls/apps/app_meetme.c
team/junky/cli-tls/apps/app_minivm.c
team/junky/cli-tls/apps/app_parkandannounce.c
team/junky/cli-tls/apps/app_queue.c
team/junky/cli-tls/apps/app_rpt.c
team/junky/cli-tls/apps/app_voicemail.c
team/junky/cli-tls/bridges/bridge_builtin_features.c
team/junky/cli-tls/build_tools/menuselect-deps.in
team/junky/cli-tls/cdr/cdr_custom.c
team/junky/cli-tls/cdr/cdr_manager.c
team/junky/cli-tls/cdr/cdr_sqlite.c
team/junky/cli-tls/cdr/cdr_sqlite3_custom.c
team/junky/cli-tls/channels/Makefile
team/junky/cli-tls/channels/chan_agent.c
team/junky/cli-tls/channels/chan_alsa.c
team/junky/cli-tls/channels/chan_bridge.c
team/junky/cli-tls/channels/chan_console.c
team/junky/cli-tls/channels/chan_dahdi.c
team/junky/cli-tls/channels/chan_gtalk.c
team/junky/cli-tls/channels/chan_h323.c
team/junky/cli-tls/channels/chan_iax2.c
team/junky/cli-tls/channels/chan_jingle.c
team/junky/cli-tls/channels/chan_local.c
team/junky/cli-tls/channels/chan_mgcp.c
team/junky/cli-tls/channels/chan_misdn.c
team/junky/cli-tls/channels/chan_nbs.c
team/junky/cli-tls/channels/chan_oss.c
team/junky/cli-tls/channels/chan_phone.c
team/junky/cli-tls/channels/chan_sip.c
team/junky/cli-tls/channels/chan_skinny.c
team/junky/cli-tls/channels/chan_unistim.c
team/junky/cli-tls/channels/chan_usbradio.c
team/junky/cli-tls/channels/chan_vpb.cc
team/junky/cli-tls/channels/sig_analog.c
team/junky/cli-tls/channels/sig_analog.h
team/junky/cli-tls/configs/chan_dahdi.conf.sample
team/junky/cli-tls/configs/sip.conf.sample
team/junky/cli-tls/configure
team/junky/cli-tls/configure.ac
team/junky/cli-tls/doc/manager_1_1.txt
team/junky/cli-tls/doc/tex/asterisk.tex
team/junky/cli-tls/funcs/func_cdr.c
team/junky/cli-tls/funcs/func_channel.c
team/junky/cli-tls/funcs/func_odbc.c
team/junky/cli-tls/include/asterisk/_private.h
team/junky/cli-tls/include/asterisk/autoconfig.h.in
team/junky/cli-tls/include/asterisk/cdr.h
team/junky/cli-tls/include/asterisk/channel.h
team/junky/cli-tls/include/asterisk/devicestate.h
team/junky/cli-tls/include/asterisk/doxyref.h
team/junky/cli-tls/include/asterisk/event.h (contents, props changed)
team/junky/cli-tls/include/asterisk/event_defs.h (contents, props changed)
team/junky/cli-tls/include/asterisk/frame.h
team/junky/cli-tls/include/asterisk/utils.h
team/junky/cli-tls/main/app.c
team/junky/cli-tls/main/ast_expr2.c
team/junky/cli-tls/main/ast_expr2.fl
team/junky/cli-tls/main/asterisk.c
team/junky/cli-tls/main/cdr.c
team/junky/cli-tls/main/channel.c
team/junky/cli-tls/main/cli.c
team/junky/cli-tls/main/devicestate.c
team/junky/cli-tls/main/dial.c
team/junky/cli-tls/main/event.c (contents, props changed)
team/junky/cli-tls/main/features.c
team/junky/cli-tls/main/file.c
team/junky/cli-tls/main/frame.c
team/junky/cli-tls/main/loader.c
team/junky/cli-tls/main/logger.c
team/junky/cli-tls/main/manager.c
team/junky/cli-tls/main/pbx.c
team/junky/cli-tls/main/rtp_engine.c
team/junky/cli-tls/res/ais/evt.c
team/junky/cli-tls/res/res_agi.c
team/junky/cli-tls/res/res_calendar.c
team/junky/cli-tls/tests/test_substitution.c
Propchange: team/junky/cli-tls/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/junky/cli-tls/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/junky/cli-tls/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jun 29 23:28:56 2009
@@ -1,1 +1,1 @@
-/trunk:1-203037
+/trunk:1-204381
Modified: team/junky/cli-tls/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/CHANGES?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/CHANGES (original)
+++ team/junky/cli-tls/CHANGES Mon Jun 29 23:28:56 2009
@@ -7,6 +7,7 @@
=== and the other UPGRADE files for older releases.
===
======================================================================
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3 -------------
------------------------------------------------------------------------------
@@ -33,6 +34,13 @@
will accept the SDP even if the SDP version number is not properly incremented,
but will generate a warning in the log indicating that the SIP peer that sent
the SDP should have the 'ignoresdpversion' option set.
+ * The 'nat' option has now been been changed to have yes, no, force_rport, and
+ comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
+ symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
+ remote side requests it and disables symmetric RTP support. Setting it to
+ force_rport forces RFC 3581 behavior and disables symmetric RTP support.
+ Setting it to comedia enables RFC 3581 behavior if the remote side requests it
+ and enables symmetric RTP support.
IAX2 Changes
-----------
@@ -42,16 +50,19 @@
Applications
------------
* Added progress option to the app_dial D() option. When progress DTMF is
- present, those values are sent immediatly upon receiving a PROGRESS message
+ present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
* Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
* Added c() option to app_chanspy. This option allows custom DTMF to be set
- to cycle through the next avaliable channel. By default this is still '*'.
+ to cycle through the next available channel. By default this is still '*'.
* Added x() option to app_chanspy. This option allows DTMF to be set to
exit the application.
* The Voicemail application has been improved to automatically ignore messages
that only contain silence.
+ * The ChanSpy application now has the 's' option, which makes the application
+ automatically exit once it hits a point where no more channels are available
+ to spy on.
Dialplan Functions
------------------
@@ -76,7 +87,7 @@
The possible values are:
- on - normal mode (the echo canceller is actually reinitalized)
+ on - normal mode (the echo canceller is actually reinitialized)
off - disabled
fax - FAX/data mode (NLP disabled if possible, otherwise completely
disabled)
@@ -155,15 +166,24 @@
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
-Logger
-------
- * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
- users of this channel in the tree have been converted to LOG_NOTICE or removed
- (in cases where the same message was already generated to another channel).
+Channel Event Logging
+---------------------
+ * A new interface, CEL, is introduced here. CEL logs single events, much like
+ the AMI, but it differs from the AMI in that it logs to db backends much
+ like CDR does; is based on the event subsystem introduced by Russell, and
+ can share in all its benefits; allows multiple backends to operate like CDR;
+ is specialized to event data that would be of concern to billing sytems,
+ like CDR. Backends for logging and accounting calls have been produced,
+ but a new CDR backend is still in development.
CDR
---
+ * 'linkedid' and 'peeraccount' are new CDR fields available to CDR officianados.
+ linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
+ etc are performed. Thus the peices of CDR can be grouped into multilegged sets.
* Multiple files and formats can now be specified in cdr_custom.conf.
+ * cdr_syslog has been added which allows CDRs to be written directly to syslog.
+ See configs/cdr_syslog.conf.sample for more information.
Calendaring for Asterisk
------------------------
@@ -172,6 +192,16 @@
as well as the ability to execute dialplan logic upon calendar event notifications.
iCalendar, CalDAV, and Exchange Server calendars are supported (Exchange support
only tested on Exchange Server 2003 with no support for forms-based authentication).
+
+Multicast RTP Support
+---------------------
+ * A new RTP engine and channel driver have been added which supports Multicast RTP.
+ The channel driver can be used with the Page application to perform multicast RTP
+ paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
+ Type can be either basic or linksys.
+ Destination is the IP address and port for the RTP packets.
+ Control address is specific to the linksys type and is used for sending the control
+ packets unique to them.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
@@ -469,12 +499,8 @@
as well as periodically updating the IP address. These properties allow for
better performance as well as recovery in the event of an IP change.
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
- load/reload of large numbers of peers/users by ~40x (for large lists of peers.
- Initially, we saw 4x improvement in call setup/destruction, but at the time
- of merging, this gain has disappeared; further research will be done to try
- and restore this performance improvement. Astobj2 refcounting is now used
- for users, peers, and dialogs. Users are encouraged to assist in regression
- testing and problem reporting!
+ load/reload of large numbers of peers/users by ~40x (for large lists of peers).
+ These changes also provide performance improvements for call setup and tear down.
* Added ability to specify registration expiry time on a per registration basis in
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
@@ -1110,11 +1136,11 @@
--------------
* Added rotatestrategy option to logger.conf, along with two new options:
"timestamp" which will use the time to name the logger files instead of
- sequence number; and "rotate", which rotates the names of the logfiles,
+ sequence number; and "rotate", which rotates the names of the log files,
similar to the way syslog rotates files.
* Added exec_after_rotate option to logger.conf, which allows a system
command to be run after rotation. This is primarily useful with
- rotatestrategry=rotate, to allow a limit on the number of logfiles kept
+ rotatestrategy=rotate, to allow a limit on the number of log files kept
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
Modified: team/junky/cli-tls/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/Makefile?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/Makefile (original)
+++ team/junky/cli-tls/Makefile Mon Jun 29 23:28:56 2009
@@ -293,7 +293,7 @@
# value directly to ASTCFLAGS
ASTCFLAGS+=$(MALLOC_DEBUG)$(OPTIONS)
-MOD_SUBDIRS:=channels pbx apps codecs formats cdr bridges funcs tests main res $(LOCAL_MOD_SUBDIRS)
+MOD_SUBDIRS:=channels pbx apps codecs formats cdr cel bridges funcs tests main res $(LOCAL_MOD_SUBDIRS)
OTHER_SUBDIRS:=utils agi
SUBDIRS:=$(OTHER_SUBDIRS) $(MOD_SUBDIRS)
SUBDIRS_INSTALL:=$(SUBDIRS:%=%-install)
@@ -574,6 +574,8 @@
mkdir -p $(DESTDIR)$(ASTDATADIR)/documentation/thirdparty
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
+ mkdir -p $(DESTDIR)$(ASTLOGDIR)/cel-csv
+ mkdir -p $(DESTDIR)$(ASTLOGDIR)/cel-custom
mkdir -p $(DESTDIR)$(ASTDATADIR)/keys
mkdir -p $(DESTDIR)$(ASTDATADIR)/firmware
mkdir -p $(DESTDIR)$(ASTDATADIR)/firmware/iax
Modified: team/junky/cli-tls/UPGRADE.txt
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/UPGRADE.txt?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/UPGRADE.txt (original)
+++ team/junky/cli-tls/UPGRADE.txt Mon Jun 29 23:28:56 2009
@@ -20,11 +20,16 @@
From 1.6.2 to 1.6.3:
+* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
+ users of this channel in the tree have been converted to LOG_NOTICE or removed
+ (in cases where the same message was already generated to another channel).
+
* The usage of RTP inside of Asterisk has now become modularized. This means
the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
If you are not using autoload=yes in modules.conf you will need to ensure
it is set to load. If not, then any module which uses RTP (such as chan_sip)
will not be able to send or receive calls.
+
* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
remains. It now exists within app_chanspy.c and retains the exact same
functionality as before.
Modified: team/junky/cli-tls/apps/app_chanisavail.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/apps/app_chanisavail.c?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/apps/app_chanisavail.c (original)
+++ team/junky/cli-tls/apps/app_chanisavail.c Mon Jun 29 23:28:56 2009
@@ -159,7 +159,7 @@
snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
status = inuse = ast_device_state(trychan);
}
- if ((inuse <= 1) && (tempchan = ast_request(tech, chan->nativeformats, number, &status))) {
+ if ((inuse <= 1) && (tempchan = ast_request(tech, chan->nativeformats, chan, number, &status))) {
ast_str_append(&tmp_availchan, 0, "%s%s", ast_str_strlen(tmp_availchan) ? "&" : "", tempchan->name);
snprintf(tmp, sizeof(tmp), "%s/%s", tech, number);
Modified: team/junky/cli-tls/apps/app_chanspy.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/apps/app_chanspy.c?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/apps/app_chanspy.c (original)
+++ team/junky/cli-tls/apps/app_chanspy.c Mon Jun 29 23:28:56 2009
@@ -134,6 +134,9 @@
</option>
<option name="o">
<para>Only listen to audio coming from this channel.</para>
+ </option>
+ <option name="s">
+ <para>Stop when no more channels are left to spy on.</para>
</option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
@@ -266,6 +269,9 @@
<option name="o">
<para>Only listen to audio coming from this channel.</para>
</option>
+ <option name="s">
+ <para>Stop when there are no more extensions left to spy on.</para>
+ </option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
numeric extension in the current context or the context
@@ -349,6 +355,7 @@
OPTION_DTMF_EXIT = (1 << 14), /* Set DTMF to exit, added for DAHDIScan integration */
OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next avaliable channel, (default is '*') */
OPTION_DAHDI_SCAN = (1 << 16), /* Scan groups in DAHDIScan mode */
+ OPTION_STOP = (1 << 17),
};
enum {
@@ -373,6 +380,7 @@
AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
AST_APP_OPTION_ARG('e', OPTION_ENFORCED, OPT_ARG_ENFORCED),
AST_APP_OPTION('o', OPTION_READONLY),
+ AST_APP_OPTION('s', OPTION_STOP),
AST_APP_OPTION('X', OPTION_EXIT),
AST_APP_OPTION('s', OPTION_NOTECH),
AST_APP_OPTION_ARG('n', OPTION_NAME, OPT_ARG_NAME),
@@ -956,6 +964,9 @@
if (res == -1 || ast_check_hangup(chan))
break;
+ if (ast_test_flag(flags, OPTION_STOP) && !next_autochan) {
+ break;
+ }
}
exit:
Modified: team/junky/cli-tls/apps/app_confbridge.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/junky/cli-tls/apps/app_confbridge.c?view=diff&rev=204382&r1=204381&r2=204382
==============================================================================
--- team/junky/cli-tls/apps/app_confbridge.c (original)
+++ team/junky/cli-tls/apps/app_confbridge.c Mon Jun 29 23:28:56 2009
@@ -559,7 +559,7 @@
if (!(conference_bridge->playback_chan)) {
int cause;
[... 12679 lines stripped ...]
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