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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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left to spy on.
(closes issue #14594)
Reported by: JimDickenson
Patches:
chanspy.diff uploaded by JimDickenson (license 710)
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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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(closes issue #14596)
Reported by: JimDickenson
Patches:
pong2.diff uploaded by JimDickenson (license 710)
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(closes issue #14452)
Reported by: umberto71
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configurable values for finer control.
(closes issue #8855)
Reported by: mikma
Tested by: klaus3000, file
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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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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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(closes issue #15404)
Reported by: leobrown
Patches:
iax_codec_debug.patch uploaded by leobrown (license 541)
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(closes issue #14413)
Reported by: pj
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application and channel.
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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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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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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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This has the pleasant side effect of cleaning up the header inclusion process
in logger.c.
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(issue #15269)
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GR-303.
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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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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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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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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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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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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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(closes issue #11797)
Reported by: macbrody
Review: https://reviewboard.asterisk.org/r/270/
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
Improved chan_dahdi.conf pritimer error checking.
Valid format is: pritimer=timer_name,timer_value
* Fixed segfault if the ',' is missing.
* Completely check the range returned by pri_timer2idx() to prevent
possible access outside array bounds.
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r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines
Use the handy UNLINK macro instead of hand-coding the same thing in-line.
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Also change the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a hostname) to
'connection' and added some verbage explaining that the user would need to
refer to their freetds.conf file for those settings. 'hostname' was kept
as a backwards compatible configuration parameter.
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setting NAI properly on linksets in certain situations.
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transfer messages.
If the outgoing_colp parameter is set to not send COLP information, then
it does not make sense to send redirecting or transfer messages announcing
new COLP information that is blocked. The service provider may supply the
listed number for that line when it passes the messages to the next hop.
Why tell the switch that these events happened when the information is
otherwise suppressed?
Also blocked the number of previous redirects that may have occurred to
calls going out the port when outgoing_colp is 2.
Follow on to JIRA ABE-1853.
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(closes issue #14509)
Reported by: timeshell
Patches:
20090504__bug14509.diff.txt uploaded by tilghman (license 14)
Tested by: awk, timeshell
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines
MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport
(closes issue #14659)
Reported by: klaus3000
Patches:
patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65)
mwi_port-transport_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, klaus3000
Review: https://reviewboard.asterisk.org/r/288/
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r202601 | mmichelson | 2009-06-23 10:22:35 -0500 (Tue, 23 Jun 2009) | 3 lines
Fix more memory leaks that may result if rtp is not successfully allocated.
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r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
Fix potential memory leak in chan_sip when video rtp is not allocated properly.
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(closes issue #2264)
Reported by: pfn
Patches:
silent-vm-1.6.2.txt uploaded by pfn (license 810)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines
Report CallerID change during a masquerade.
Reported by: markster
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Pointed out by Russell while working on the CEL branch.
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