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Reported by philipp64 in #asterisk-dev.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252623 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16849)
Reported by: ip-rob
Patches:
20100311__issue16849.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251989 f38db490-d61c-443f-a65b-d21fe96a405b
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quotes during parsing.
(closes issue #16905)
Reported by: ip-rob
Patches:
20100303__issue16905.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251884 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251877 f38db490-d61c-443f-a65b-d21fe96a405b
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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:
directory_pause.trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/481/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251779 f38db490-d61c-443f-a65b-d21fe96a405b
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For some reason the documentation for the 'k' application in trunk
and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them all
to match. The wording in 1.6.2 and trunk was ambiguous, so you could
interpret the wording the mean that recording would continue upon hangup
indefinitely, or you could interpret it to mean that the recorded
data would not be discarded upon hangup. This change makes it clear
we mean the latter, and not the former.
Came from a discussion in #asterisk on IRC.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251680 f38db490-d61c-443f-a65b-d21fe96a405b
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Fixes regression introduced in 140167 that uses the wrong variable names.
(closes issue #16930)
Reported by: ianc
Patches:
fix_reload_followme.diff uploaded by ianc (license 998)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250979 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16953)
Reported by: elguero
Patches:
app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250913 f38db490-d61c-443f-a65b-d21fe96a405b
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Only chan_dahdi set a value in cdrflags. Everyone else just copied it
around the system. Noone cared about any value it may have contained.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
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Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250302 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250141 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16145)
Reported by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249950 f38db490-d61c-443f-a65b-d21fe96a405b
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caused a regression, as only supported VOICE, not VIDEO etc.
(issue #16880)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249947 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249892 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249891 f38db490-d61c-443f-a65b-d21fe96a405b
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VMSayName that will play the recorded name of the voicemail user if it exists,
otherwise will play the mailbox number. A unit test has been written to verify
correct functionality called test_voicemail_vmsayname.
(closes issue #14973)
Reported by: ghjm
Review: https://reviewboard.asterisk.org/r/530/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249889 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:
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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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249623 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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- 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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(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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r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines
Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247169 f38db490-d61c-443f-a65b-d21fe96a405b
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targets to be more particular.
This change builds upon the recent change to menuselect to add 'touch_on_change'
as an attribute of both categories and members. This should allow only the most
invasive defines to cause a complete rebuild, while defines which only affect a
subset of modules will only cause a rebuild of that smaller set.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246789 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246382 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines
fixes random deadlock in app_queue with use_weight during reload
(closes issue #16677)
Reported by: tim_ringenbach
Patches:
app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246116 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16361)
Reported by: vlad
Patches:
20100208__issue16361.diff.txt uploaded by tilghman (license 14)
Tested by: kenny, bloodoff, misaksen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245729 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245680 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel.
(closes issue #16687)
Reported by: bklang
Patches:
app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919)
(with modifications)
(closes issue #16686)
Reported by: bklang
Patches:
app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
(with modifications)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244393 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01 Feb 2010) | 11 lines
Backup and restore original textfile, for prosthesis (gerund of prepend).
Also, fix menuselect such that changing voicemail build options correctly
causes rebuild.
(closes issue #16415)
Reported by: tomo1657
Patches:
prepention.patch uploaded by tomo1657 (license 484)
(with modifications by me to backport to 1.4)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244243 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines
Revert 243570, I should have looked at this closer. Will reopen the issue, but
am leaving the review closed as the change was pointless.
(issue #16488)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243693 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines
Extend announcement URL used with Queue from 80 chars to PATH_MAX.
(closes issue #16488)
Reported by: syspert
Patches:
soundfilelen.pacth-2 uploaded by syspert (license 938)
Review: https://reviewboard.asterisk.org/r/475/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243571 f38db490-d61c-443f-a65b-d21fe96a405b
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Pushes code clean up done in app_externalivr back
into app_senddtmf
Review: https://reviewboard.asterisk.org/r/473/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243346 f38db490-d61c-443f-a65b-d21fe96a405b
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Implemented a new command 'D' that allows client
IVRs to send DTMF digits to the channel.
(closes issue #16615)
Reported by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/465/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242357 f38db490-d61c-443f-a65b-d21fe96a405b
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See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241364 f38db490-d61c-443f-a65b-d21fe96a405b
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Add file information to data element of T event so
the file information is sent to the client when it is
interrupted. Previously only notification of pending
files that were dropped was sent
(closes issue #16147)
Reported by: thedavidfactor
Tested by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/449/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240969 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240842 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16431)
Reported by: syspert
Patches:
20100112__issue16431.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240421 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240419 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240415 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240368 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239712 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239624 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16078)
Reported by: RoadKill
Patches:
quequename_limit.patch uploaded by ppyy (license 906)
Tested by: dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238361 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15725)
Reported by: shanermn
Patches:
v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238181 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines
Resolve a crash due to an ast_frame not being fully initialized.
(closes issue #16531)
Reported by: john8675309
(closes SWP-615)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238010 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237920 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237882 f38db490-d61c-443f-a65b-d21fe96a405b
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