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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252617 f38db490-d61c-443f-a65b-d21fe96a405b
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Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252533 f38db490-d61c-443f-a65b-d21fe96a405b
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Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252531 f38db490-d61c-443f-a65b-d21fe96a405b
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Reviewboard: https://reviewboard.asterisk.org/r/551/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@252361 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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ParkAndAnnounce
(closes issue #16731)
Reported by: sebele67
Patches:
issue16731_20100129.diff uploaded by seanbright (license 71)
Tested by: sebele67
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251410 f38db490-d61c-443f-a65b-d21fe96a405b
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When using the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so I'm
implementing the change.
(closes issue #16784)
Reported by: pabelanger
Tested by: pabelanger, mnick, davidw, mutineer612
(closes issue #16887)
Reported by: jlpedrosa
Tested by: jlpedrosa, mutineer612
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@251309 f38db490-d61c-443f-a65b-d21fe96a405b
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Don't attempt to chdir on a URL!
(closes issue #16875)
Reported by: raarts
Patches:
moh-http.patch uploaded by raarts (license 937)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250786 f38db490-d61c-443f-a65b-d21fe96a405b
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A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.
(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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250613 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250480 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250394 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
(closes issue #16704)
Reported by: TimeHider
Tested by: lmadsen, TimeHider
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250050 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16267)
Reported by: elsto
Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10)
Tested by: davidw, elsto
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250043 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #16855)
Reported by: davidw
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@250041 f38db490-d61c-443f-a65b-d21fe96a405b
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caused a regression, as only supported VOICE, not VIDEO etc.
Left in small formatting change.
(issue #16880)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249946 f38db490-d61c-443f-a65b-d21fe96a405b
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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:
based on 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/1.4@249845 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249671 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249536 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:
based on overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
(closes issue #16789)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249365 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249234 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16792)
Reported by: vrban
Patches:
t38_606.patch uploaded by vrban (license 756)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@249100 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248860 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248859 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248757 f38db490-d61c-443f-a65b-d21fe96a405b
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This same patch was merged in 220833, but was skipped in this branch
erroneously.
(closes issue #16170)
Reported by: francesco_r
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248668 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16786)
Reported by: dodo
Patches:
logger2.patch uploaded by dodo (license 989)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248582 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16862)
Reported by: pwalker
Patches:
replaces_deadlock_1.4 uploaded by dvossel (license 671)
Tested by: pwalker, dvossel
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crash Asterisk.
(closes issue #16470)
Reported by: kjotte
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248012 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247910 f38db490-d61c-443f-a65b-d21fe96a405b
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already have one.
(closes issue #16331)
Reported by: bluefox
Tested by: mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247651 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #16808)
Reported by: lmadsen
(issue #16810)
Reported by: Nick_Lewis
Tested by: lmadsen
Review: https://reviewboard.asterisk.org/r/507/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247502 f38db490-d61c-443f-a65b-d21fe96a405b
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1) Fix the check to see if we are using wget to not be full of fail. The
configure script populates this variable with the absolute path to wget if
it is found, so it didn't work.
2) Allow some extra arguments to be passed in for wget. This is just a simple
change to allow our Bamboo build script to tell wget to be quiet and not fill
up our logs with download status output.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247422 f38db490-d61c-443f-a65b-d21fe96a405b
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the queue cannot place calls.
(closes issue #16834)
Reported by: kebl0155
Patches:
app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@247168 f38db490-d61c-443f-a65b-d21fe96a405b
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actually solve dependency problems.
(Previously, it would fail out again with the same message about running
'make menuselect', which was NOT at all helpful.)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246709 f38db490-d61c-443f-a65b-d21fe96a405b
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On channel destruction the channel's datastores are removed and
destroyed. Since there are public API calls to find and remove
datastores on a channel, a lock should be held whenever datastores are
removed and destroyed. This resolves a crash caused by a race
condition in app_chanspy.c.
(closes issue #16678)
Reported by: tim_ringenbach
Patches:
datastore_destroy_race.diff uploaded by tim ringenbach (license 540)
Tested by: dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246545 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16677)
Reported by: tim_ringenbach
Patches:
app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@246115 f38db490-d61c-443f-a65b-d21fe96a405b
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risk.
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depends on res_smdi
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@245909 f38db490-d61c-443f-a65b-d21fe96a405b
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2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
(closes issue #15997)
Reported by: exarv
Patches:
iax_fix.diff uploaded by dvossel (license 671)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@245792 f38db490-d61c-443f-a65b-d21fe96a405b
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People don't always build Asterisk from source (distro packages, anybody?).
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Remove explicit license for IAXy firmware as it is no longer included in the tree
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@245044 f38db490-d61c-443f-a65b-d21fe96a405b
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