Age | Commit message (Collapse) | Author | Files | Lines |
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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.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249801 f38db490-d61c-443f-a65b-d21fe96a405b
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regression since sig_pri split.
(issue #16909)
Reported by: alecdavis
Patches:
pritimer.asterisk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249759 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249672 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch splits transmit_displaymessage into transmit_clear_display_message and transmit_display_message which better aligns with the skinny protocol. The new transmit_display_message is not used in the current code, but will be and so it is commented.
Moved handle_datetime from this function to onhook and offhook functions (display now properly cleared at the end of a call on 30VIP).
Removed skinny debug messages from inline code as there's an ast_verb in transmit_clear_display_message. Also, removed commentary that it was a clear display as it is now apparent from the function name.
Split transmit_displaypromptmessage into display and clear.
(closes issue #16878)
Reported by: wedhorn
Patches:
skinny-clean02.diff uploaded by wedhorn (license 30)
skinny-clean03.diff uploaded by wedhorn (license 30)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249670 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16826)
Reported by: PipoCanaja
Patches:
chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja (license 994)
Tested by: wedhorn
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249669 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249623 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249538 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16927)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249491 f38db490-d61c-443f-a65b-d21fe96a405b
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previous test, gave false level of assurance that code was healthy.
(issue #16927)
Reported by: alecdavis
Patches:
based on app_voicemail_test.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249449 f38db490-d61c-443f-a65b-d21fe96a405b
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the -dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249405 f38db490-d61c-443f-a65b-d21fe96a405b
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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/trunk@249320 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249235 f38db490-d61c-443f-a65b-d21fe96a405b
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- 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.
- Unit tests added to verify behavior in the future.
(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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249187 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249186 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16769)
Reported by: syspert
Patches:
bridgeaction.patch uploaded by syspert (license 938)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249105 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249101 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249050 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249013 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249012 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249011 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249010 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249009 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248952 f38db490-d61c-443f-a65b-d21fe96a405b
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AST-2010-003
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248946 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248861 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248793 f38db490-d61c-443f-a65b-d21fe96a405b
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r248668 | jpeeler | 2010-02-24 16:58:54 -0600 (Wed, 24 Feb 2010) | 8 lines
Make deletion of temporary greetings work properly with IMAP_STORAGE
This same patch was merged in 220833, but was skipped in this branch
erroneously.
(closes issue #16170)
Reported by: francesco_r
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248671 f38db490-d61c-443f-a65b-d21fe96a405b
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when we
change flags. Fixes a weird symbol problem mmichelson was having in a group branch,
but also applies to trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248667 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248584 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248534 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248533 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/517
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248489 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248397 f38db490-d61c-443f-a65b-d21fe96a405b
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Uncommenting the REF_DEBUG definition where it was in the source
resulted in only a small part of the astobj2 references being logged
to a file. Moving this up higher in the include list causes all references
to be logged as they should be.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248347 f38db490-d61c-443f-a65b-d21fe96a405b
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r248268 | oej | 2010-02-22 14:52:34 +0100 (Mån, 22 Feb 2010) | 2 lines
Don't log to debug unless debug is turned on
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248269 f38db490-d61c-443f-a65b-d21fe96a405b
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Fix the copyright lines, tweak doxygen formatting, and remove some unnecessary
includes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248226 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248225 f38db490-d61c-443f-a65b-d21fe96a405b
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Break transmit_tone into transmit_start_tone and transmit_stop_tone as per the skinny protocol.
(closes issue #16874)
Reported by: wedhorn
Patches:
skinny-clean01.diff uploaded by wedhorn (license 30)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248184 f38db490-d61c-443f-a65b-d21fe96a405b
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r248106 | oej | 2010-02-20 23:25:42 +0100 (Lör, 20 Feb 2010) | 2 lines
Make sure we support RTCP compound messages with zero reports
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248107 f38db490-d61c-443f-a65b-d21fe96a405b
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mfcr2_immediate_accept work again, reported and patched by korihor
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248003 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247915 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247914 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16368)
Reported by: thirionjwf
Patches:
res_speech.c.patch uploaded by thirionjwf (license 955)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247841 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247787 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16471)
Reported by: kjotte
Patches:
M16471.diff uploaded by junky (license 177)
Tested by: kjotte, junky
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247770 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16475)
Reported by: okrief
Patches:
queue_crash.diff uploaded by dvossel (license 671)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247736 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines
Copy the calling party's account code to the called party if they don't already have one.
(closes issue #16331)
Reported by: bluefox
Tested by: mnicholson
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