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r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) | 11 lines
Update existing Local channel documentation.
A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.
(closes issue #16637)
Reported by: kobaz
Patches:
localchannel.tex uploaded by lmadsen (license 10)
localchannel.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jsmith, mmichelson
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r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines
Merged revisions 250480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines
Make sure to clear red alarm after polarity reversal.
From the issue:
The automatic overnight line tests (or manual ones) used on UK (BT) lines causes
a red alarm on a dahdi / TDM400P connected channel. This is because the line
uses voltage tests (battery loss) and polarity reversal. The polarity reversal
causes chan_dahdi to initiate v23 CallerID processing but during this the event
DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared.
(closes issue #14163)
Reported by: jedi98
Patches:
chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653)
Tested by: mattbrown, Chainsaw, mikeeccleston
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r250478 | dvossel | 2010-03-03 13:02:13 -0600 (Wed, 03 Mar 2010) | 2 lines
Changes 0ms to <1ms in cli END results during 'test execute'
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r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines
Merged revisions 250394 via svnmerge from
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r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines
fixes problem with duplicate TXREQ packets
When Asterisk receives an IAX2 TXREQ packet, try_transfer()
will call store_by_transfercallno() to link the chan_iax2_pvt
struct into iax_transfercallno_pvts. If a duplicate TXREQ
packet is received for the same call, the pvt struct will be
linked into iax_transfercallno_pvts multiple times. This patch
fixes this. Thanks rain for debugging this and providing a patch!
(closes issue #16904)
Reported by: rain
Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327)
Tested by: rain, dvossel
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r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
fixes signed to unsigned int comparision issue for FaxMaxDatagram value.
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r250237 | dvossel | 2010-03-02 18:04:28 -0600 (Tue, 02 Mar 2010) | 2 lines
fixes assumption that test failed if it did not pass when generating results
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r250235 | dvossel | 2010-03-02 17:38:29 -0600 (Tue, 02 Mar 2010) | 2 lines
base64 unit test
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r250141 | dvossel | 2010-03-02 15:58:03 -0600 (Tue, 02 Mar 2010) | 15 lines
adds 'p' option to PickupChan
The 'p' option allows the PickupChan app to pickup
a ringing phone by looking for the first match to a
partial channel name rather than requiring a full match.
(closes issue #16613)
Reported by: syspert
Patches:
pickipbycallid.patch uploaded by syspert (license 938)
pickupbycallerid_v2.patch uploaded by dvossel (license 671)
Tested by: dvossel, syspert
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r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines
Update IMAP documentation.
Update the IMAP documentation to make it clear that storing voicemails
in the same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.
(issue #16704)
Reported by: TimeHider
Tested by: lmadsen, TimeHider
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r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines
Merged revisions 250043 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines
Update documentation to clarify purpose of unanswered option.
(closes issue #16267)
Reported by: elsto
Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10)
Tested by: davidw, elsto
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r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines
Update documentation to not imply we support overriding options.
(closes issue #16855)
Reported by: davidw
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r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) | 4 lines
Fix literal values wrapped in documentation.
(closes issue #16145)
Reported by: tilghman
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caused a regression, as only supported VOICE, not VIDEO etc.
(issue #16880)
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r249925 | lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
Add missing description of the PARKINGLOT variable in XML documentation.
(closes issue #16743)
Reported by: snuffy
Patches:
parkingdoc.diff uploaded by snuffy (license 35)
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r249912 | lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
Convert some DUNDI functions to XML documentation.
(closes issue #16798)
Reported by: snuffy
Patches:
xml_dundi.diff uploaded by snuffy (license 35)
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r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration
When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
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r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) | 1 line
Fix several XML documentation validate errors.
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249844 f38db490-d61c-443f-a65b-d21fe96a405b
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r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines
Merged revisions 249671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines
Fix crash in app_voicemail related to message counting.
We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *'
causing a segfault.
(closes issue #16921)
Reported by: whardier
Patches:
20100301_issue16921.patch uploaded by seanbright (license 71)
Tested by: whardier
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r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) | 2 lines
Constify a bit of app_voicemail, to make ODBC and IMAP compile once again.
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r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines
Merged revisions 249536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines
Modify queued frames from local channels to not set the other side to up
In this case, attended transfers were broken due to ast_feature_request_and_dial
detecting the channel being set to up before the answer frame could be read and
therefore failing to mark the channel as ready. This fix is a regression fix for
244785, which should continue to work properly as well.
(closes issue #16816)
Reported by: jamhed
Tested by: jamhed, corruptor
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r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) | 5 lines
Fix unit test that Alec Davis broke.
(closes issue #16927)
Reported by: alecdavis
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r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
Properly document voicemail API documents. Also fix a crash reported via the -dev list.
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Following Q.931 5.2.4
When the user has determined that sufficient call information has been received the
user shall stop T302 and send CALL PROCEEDING to the network.
Previously timeouts were possible if the dialplan took a long time to issue any
response back to the network.
Verified that our local TELCO also does the same.
(issue #16789)
Reported by: alecdavis
Patches:
overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249321 f38db490-d61c-443f-a65b-d21fe96a405b
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r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines
Merged revisions 249234 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line
add a reference to the now-published IAX2 RFC
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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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r249186 | dvossel | 2010-02-26 12:41:36 -0600 (Fri, 26 Feb 2010) | 2 lines
adds Time field to "test show results" cli command
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r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines
Merged revisions 249100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines
For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792)
Reported by: vrban
Patches:
t38_606.patch uploaded by vrban (license 756)
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r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines
Merged revisions 248860 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines
Ensure that monitor recordings are written to the correct location (again)
This is an extension to 248757. As such the dialplan test has been extended:
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
exten => 5040, n, dial(sip/5001)
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
exten => 5041, n, dial(sip/5001)
exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001)
exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m)
exten => 5043, n, changemonitor(monitor_test4)
exten => 5043, n, dial(sip/5001)
exten => 5044, 1, monitor(wav,monitor_test4,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning
exten => 5044, n, dial(sip/5001)
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r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines
Fix incorrect ACL behavior when CIDR notation of "/0" is used.
AST-2010-003
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r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines
Merged revisions 248859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines
Some platforms clear /var/run at boot, which makes connecting a remote console... difficult.
Previously, we only created the default /var/run/asterisk directory at install
time. While we could create it in the init script, that would not work for
those who start asterisk manually from the command line. So the safest thing
to do is to create it as part of the Asterisk boot process. This also changes
the ownership of the directory, because the pid and ctl files are created after
we setuid/setgid.
(closes issue #16802)
Reported by: Brian
Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
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r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines
Merged revisions 248757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines
Ensure that monitor recordings are written to the correct location.
Recordings should be placed in the monitor directory when a non-absolute path
is used.
Exact dialplan used for testing:
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
exten => 5040, n, dial(sip/5001)
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
exten => 5041, n, dial(sip/5001)
exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001)
ABE-2101
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r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines
Merged revisions 248582 via svnmerge from
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r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines
Remove color code sequences from verbose messages that go to logfiles.
(closes issue #16786)
Reported by: dodo
Patches:
logger2.patch uploaded by dodo (license 989)
Tested by: tilghman
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r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines
Merged revisions 248396 via svnmerge from
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r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines
fixes invite with replaces deadlock
(closes issue #16862)
Reported by: pwalker
Patches:
replaces_deadlock_1.4 uploaded by dvossel (license 671)
Tested by: pwalker, dvossel
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r248108 | oej | 2010-02-20 23:37:22 +0100 (Lör, 20 Feb 2010) | 2 lines
Improve support for RTCP reports without report blocks
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r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines
Fix various problems detected with Valgrind.
* chan_console accessed pvts after deallocation.
* The module loader did not check usecount on shutdown, which led to chan_iax2
reading a timer that was already unloaded.
(closes issue #16062)
Reported by: alexanderheinz
Patches:
20091109__issue16062.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 line
mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor
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r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines
Merged revisions 247910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines
Merged revision 247904 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
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r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines
Make chan_misdn DTMF processing consistent with other channel technologies.
The processing of DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels, especially
DAHDI analog. This causes DTMF tones sent from an ISDN phone to be
doubled at the connected party.
We are using the following 2 options of misdn.conf
1) astdtmf=yes
2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection is better
than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when mISDN
channels are connected to SIP channels with out of band DTMF for example.
The symptom is that DTMF tones sent by an ISDN phone are doubled on the
way through asterisk when two mISDN channels are connected with a Local
channel in between or if it is bridged to an analog channel.
The doubling of DTMF tones is because DTMF is passed inband to asterisk by
the mISDN channel and passed out of band once again after the release of
the DTMF tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF. Analog and
SIP channels filter out the DTMF tones because they use the voice frames
returned by ast_dsp_process. But chan_misdn passes the unfiltered input
voice frames instead.
To overcome one aspect of the problem, the doubling of DTMF tones when two
mISDN channels are directly bridged, someone made an 'optimization', where
in that case the DTMF tone passed out-of-band to the peer channel is not
translated to an inband tone at the transmit side. This optimization is
bad because it does not work in general. For example, analog channels or
mISDN channels when bridged through an intermediary local channel will
generate DTMF tones from out-of-band information. Also, of course, it
must not be done when there is no inband DTMF available.
This patch fixes the issue. Now chan_misdn will filter the received
inband DTMF signal the same as other channel types.
Another change included: No need to build an extra translation path
because ast_process_dsp does it if required.
Patches:
misdn-dtmf.patch
JIRA ABE-2080
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r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines
handle_request_invite revise comment, fix coding guideline issues
I'm working with this code right now trying to analyze a deadlock.
This change is just to clean up a few things before I make a more
complex patch.
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r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
Revert an errant part of a previous cleanup, to fix a memory corruption issue.
(closes issue #16368)
Reported by: thirionjwf
Patches:
res_speech.c.patch uploaded by thirionjwf (license 955)
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r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines
If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns.
NULL means the value is not specified for the column, which normally means
the driver uses whatever is the default value. However, on MySQL, placing
a NULL in either a float or integer column results in a retrieval of the 0
value. Hence, users get an errant error on load. This patch suppresses
that error and makes the value as if it was not there.
Note that this cannot be done in the realtime driver, because the lack of
difference between NULL and 0 can only be intepreted correctly by the
driver itself. If we did it in the realtime driver, then it would be
effectively impossible to set any realtime field to 0, because it would act
as if the field were unspecified and possibly take on a different value.
(closes issue #16683)
Reported by: wdoekes
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r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) | 9 lines
fixes confbridge crash when no timing module is loaded.
(closes issue #16471)
Reported by: kjotte
Patches:
M16471.diff uploaded by junky (license 177)
Tested by: kjotte, junky
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