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r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010) | 17 lines
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r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines
fixes app_meetme dsp error
We attempted to detect silence after translating a frame
from signed linear. This caused a flooding of errors. To
resolve this the code to detect silence was moved before the
translation.
(closes issue #17133)
Reported by: jsdyer
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r261736 | jpeeler | 2010-05-06 15:11:53 -0500 (Thu, 06 May 2010) | 15 lines
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r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines
Only allow the operator key to be accepted after leaving a voicemail.
Or rather disallow the operator key from being accepted when not offered,
such as after finishing a recording from within the mailbox options menu.
ABE-2121
SWP-1267
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r260924 | jpeeler | 2010-05-04 13:51:28 -0500 (Tue, 04 May 2010) | 18 lines
Merged revisions 260923 via svnmerge from
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r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines
Voicemail transfer to operator should occur immediately, not after main menu.
There were two scenarios in the advanced options that while using the
operator=yes and review=yes options, the transfer occurred only after exiting
the main menu (after sending a reply or leaving a message for an extension).
Now after the audio is processed for the reply or message the transfer occurs
immediately as expected.
ABE-2107
ABE-2108
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r258433 | jpeeler | 2010-04-21 16:56:09 -0500 (Wed, 21 Apr 2010) | 15 lines
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r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines
Fix looping forever when no input received in certain voicemail menu scenarios.
Specifically, prompting for an extension (when leaving or forwarding a message)
or when prompting for a digit (when saving a message or changing folders).
ABE-2122
SWP-1268
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r258065 | jpeeler | 2010-04-20 12:06:19 -0500 (Tue, 20 Apr 2010) | 17 lines
Merged revisions 258029 via svnmerge from
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r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines
Play correct prompt when voicemail store failure occurs after attempted forward.
If a user's mailbox was full and a message was attempted to be forwarded to
said box, warnings on the console would indicate failure. However, the played
prompt was that of success (vm-msgsaved). Now storage failure is taken into
account and the correct prompt (vm-mailboxfull) is played when appropriate.
ABE-2123
SWP-1262
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r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines
Merged revisions 257686 via svnmerge from
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r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines
Make the mixmonitor thread process audio frames faster
Mantis issue 17078 reports MixMonitor recordings have shorter durations than
the call duration. This was because the mixmonitor thread was not processing
frames from the audiohook fast enough. The mixmonitor thread would slowly fall
behind the most recent audio frame and when the channel hangs up, the mixmonitor
thread would exit without processing the same number of frames as the channel;
leaving the mixmonitor recording shorter than actual call duration.
This revision fixes this issue by moving the ast_audiohook_trigger_wait() and
the subsequent audiohook.status check into the block where the
ast_audiohook_read_frame() function returns NULL.
(closes issue #17078)
Reported by: geoff2010
Patches:
dw-M17078.patch uploaded by dhubbard (license 733)
Tested by: dhubbard, geoff2010
Review: https://reviewboard.asterisk.org/r/611/
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r255592 | tilghman | 2010-03-31 14:13:02 -0500 (Wed, 31 Mar 2010) | 22 lines
Recorded merge of revisions 255591 via svnmerge from
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r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines
Ensure line terminators in email are consistent.
Fixes an issue with certain Mail Transport Agents, where attachments are not
interpreted correctly.
(closes issue #16557)
Reported by: jcovert
Patches:
20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: ebroad, zktech
Reviewboard: https://reviewboard.asterisk.org/r/544/
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r251877 | tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
If the argument to the system application is quoted, ensure we remove the quotes before trying to execute.
(closes issue #16842)
Reported by: ip-rob
Patches:
20100310__issue16842.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
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Since 1.6.1 CLI help reports that option p(n) 'initial pause' is available.
Supporting code was never implemented.
(closes issue #16751)
Reported by: alecdavis
Patches:
based on directory_pause.trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/481/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@251819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when called from a ISDN channel, null frames prevent '#' exit.
Now only echo back VOICE and DTMF frames
(issue #16880)
Reported by: alecdavis
Patches:
echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) | 18 lines
Cleanups to fix bugs in the VM count API functions.
- Urgent voicemails were not attached, because the attachment code looked in the wrong folder.
- Urgent voicemails were sometimes counted twice when displaying the count of new messages.
- Backends were inconsistent as to which voicemails each API counted.
(closes issue #15654)
Reported by: tomo1657
Patches:
20100225__issue15654.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
(closes issue #16448)
Reported by: hevad
Review: https://reviewboard.asterisk.org/r/525/
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r245680 | kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 lines
Don't offer MMR or JBIG transcoding during T.38 negotiation.
After further discussion with Steve Underwood, we should not (yet) be offering
to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp
release will support those features, and then they can be enabled during
negotiation
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r244393 | tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 lines
Properly respect GOSUB_RESULT as to what to do with the master channel.
Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
(closes issue #16686)
Reported by: bklang
Patches:
app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
(with modifications)
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r240415 | tilghman | 2010-01-15 14:54:24 -0600 (Fri, 15 Jan 2010) | 22 lines
Merged revisions 240414 via svnmerge from
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r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 Jan 2010) | 15 lines
Disallow leaving more than maxmsg voicemails.
This is a possibility because our previous method assumed that no messages are
left in parallel, which is not a safe assumption. Due to the vmu structure
duplication, it was necessary to track in-process messages via a separate
structure. If at some point, we switch vmu to an ao2-reference-counted
structure, which would eliminate the prior noted duplication of structures,
then we could incorporate this new in-process structure directly into vmu.
(closes issue #16271)
Reported by: sohosys
Patches:
20100108__issue16271.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14)
20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: jsutton
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r239712 | dvossel | 2010-01-13 10:31:14 -0600 (Wed, 13 Jan 2010) | 24 lines
add silence gen to wait apps
asterisk.conf's 'transmit_silence' option existed before
this patch, but was limited to only generating silence
while recording and sending DTMF. Now enabling the
transmit_silence option generates silence during wait
times as well.
To achieve this, ast_safe_sleep has been modified to
generate silence anytime no other generators are present
and transmit_silence is enabled. Wait apps not using
ast_safe_sleep now generate silence when transmit_silence
is enabled as well.
(closes issue #16524)
Reported by: kobaz
(closes issue #16523)
Reported by: kobaz
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/456/
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r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
fixes holdtime playback issue in app_queue
When reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123 seconds"
rather than "2 minutes 3 seconds".
Also, the "minute" sound file is missing, so for the moment until
that file can be created the "minutes" file is used instead.
(closes issue #16168)
Reported by: nickilo
Patches:
patch-unified-trunk-rev-222176 uploaded by nickilo (license )
Tested by: nickilo, wonderg
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r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines
Merged revisions 236509 via svnmerge from
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r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in
meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
crash, so instead just use a single fixed buffer.
(closes issue #16509)
Reported by: Kashif Raza
Patches:
20091223_16509.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r235010 | kpfleming | 2009-12-15 08:35:46 -0600 (Tue, 15 Dec 2009) | 5 lines
spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)
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r234380 | jpeeler | 2009-12-11 17:17:09 -0600 (Fri, 11 Dec 2009) | 18 lines
Merged revisions 234379 via svnmerge from
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r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
Fix talking detection status after conference user is muted.
This patch ensures that when a conference user is muted that the accompanying
AMI Meetme talking off event is sent. Also, the meetme list output is updated
to show the muted user as unmonitored.
(closes issue #16247)
Reported by: dimas
Patches:
v3-16247.patch uploaded by dimas (license 88)
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David Vossel
Tilghman Lesher
Jeff Peeler
Dwayne M. Hubbard
Leif Madsen
Russell Bryant
Sean Bright
Alec L Davis
Mark Michelson
Kevin P. Fleming