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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
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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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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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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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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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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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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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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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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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r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines
Merged revisions 204300 via svnmerge from
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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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r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines
Merged revisions 204243,204246 via svnmerge from
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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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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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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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r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines
Merged revisions 203908 via svnmerge from
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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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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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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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r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines
Merged revisions 203785 via svnmerge from
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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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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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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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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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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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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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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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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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r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines
Merged revisions 203375 via svnmerge from
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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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r203304 | jpeeler | 2009-06-25 14:54:12 -0500 (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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r203258 | qwell | 2009-06-25 14:22:46 -0500 (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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