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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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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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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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(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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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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am leaving the review closed as the change was pointless.
(issue #16488)
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(closes issue #16488)
Reported by: syspert
Patches:
soundfilelen.pacth-2 uploaded by syspert (license 938)
Review: https://reviewboard.asterisk.org/r/475/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@243570 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@240414 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 0016524)
Reported by: kobaz
(closes issue 0016523)
Reported by: kobaz
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/456/
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(closes issue #16531)
Reported by: john8675309
(closes SWP-615)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238009 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@236509 f38db490-d61c-443f-a65b-d21fe96a405b
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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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Update the documentation in ChanSpy and ExtenSpy to reflect that only a
single group can be specified to the g() option.
(closes issue #16420)
Reported by: diatonic
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@234094 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16291)
Reported by: wdoekes
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233116 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, change the use of language codes so that language registers as a prefix,
rather than an exact match.
(closes issue #16272)
Reported by: patrol-cz
Patches:
20091203__issue16272.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232820 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16369)
Reported by: vrban
Patches:
queue_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232444 f38db490-d61c-443f-a65b-d21fe96a405b
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overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239)
Reported by: CGMChris
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232355 f38db490-d61c-443f-a65b-d21fe96a405b
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app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231614 f38db490-d61c-443f-a65b-d21fe96a405b
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In app_queue, it is possible for a call_queue to be destroyed
while another object still holds a pointer to it. This patch
converts call_queue objects to ao2 objects allowing them to be
ref counted. This makes it safe for the queue_ent object in
queue_exec() to reference it's parent call_queue even after it
has left the queue.
(closes issue #15686)
Reported by: Hatrix
Patches:
v2_queue_ao2.diff uploaded by dvossel (license 671)
Tested by: dvossel, aragon
Review: https://reviewboard.asterisk.org/r/427/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231437 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16193)
Reported by: asgaroth
Patches:
bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231235 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16152)
Reported by: AlexMS
Patches:
stopmixmonitor_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, AlexMS
Review: https://reviewboard.asterisk.org/r/424/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230508 f38db490-d61c-443f-a65b-d21fe96a405b
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a workaround for it that does not change existing behavior.
(closes issue #14426)
Reported by: macli
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229965 f38db490-d61c-443f-a65b-d21fe96a405b
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hangups while playing back announcements.
(closes issue #16005)
Reported by: falves11
Patches:
dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, falves11
Review: https://reviewboard.asterisk.org/r/407/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227827 f38db490-d61c-443f-a65b-d21fe96a405b
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if the caller hung up
while the called party had not yet answered.
This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
(closes issue #14674)
Reported by: ulogic
Patches:
bug14674.patch uploaded by jpeeler (license 325)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226889 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16103)
Reported by: majorbloodnok
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enabled that should prevent it.
(closes issue #14763)
Reported by: cupotka
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224565 f38db490-d61c-443f-a65b-d21fe96a405b
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While waiting for an answer, don't send progress for branched calls
for which ringing was sent.
(closes issue #15028)
Reported by: fnordian
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223804 f38db490-d61c-443f-a65b-d21fe96a405b
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See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222152 f38db490-d61c-443f-a65b-d21fe96a405b
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chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965)
Reported by: atis
Patches:
chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96)
Tested by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@220907 f38db490-d61c-443f-a65b-d21fe96a405b
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Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
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new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@219816 f38db490-d61c-443f-a65b-d21fe96a405b
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the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218730 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218577 f38db490-d61c-443f-a65b-d21fe96a405b
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again.
(issue #15055, SWP-129)
Reported by: jthurman
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(closes issue #15100)
Reported by: lmsteffan
Patches:
(modified) pickup.patch uploaded by lmsteffan (license 779)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@218223 f38db490-d61c-443f-a65b-d21fe96a405b
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answer.
(Fixes AST-228)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@217989 f38db490-d61c-443f-a65b-d21fe96a405b
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conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@217156 f38db490-d61c-443f-a65b-d21fe96a405b
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media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@216430 f38db490-d61c-443f-a65b-d21fe96a405b
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In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@215270 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15699)
Reported by: edantie
Patches:
mixmonitor.patch uploaded by edantie (license 862)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@213103 f38db490-d61c-443f-a65b-d21fe96a405b
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The additional checks prevent generation of false TRANSFER events in certain situations.
(closes issue #14536)
Reported by: aragon
Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@211953 f38db490-d61c-443f-a65b-d21fe96a405b
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(ABE-1936)
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