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https://origsvn.digium.com/svn/asterisk/trunk
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r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line
Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@283123 f38db490-d61c-443f-a65b-d21fe96a405b
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A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@280448 f38db490-d61c-443f-a65b-d21fe96a405b
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If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279945 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17080)
Reported by: sybasesql
Patches:
20100721__issue17080.diff.txt uploaded by tilghman (license 14)
Tested by: sybasesql
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278981 f38db490-d61c-443f-a65b-d21fe96a405b
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If a channel involved in a bridge was using SLIN audio, then translation
paths were not guaranteed to be set up properly since in all likelihood
the number of translation steps was only 1.
This patch enforces the transcode_via_slin behavior if transcode_via_slin
or generic_plc is enabled and one of the formats to make compatible is
SLIN.
AST-352
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(Fixes ABE-2110)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278167 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@278023 f38db490-d61c-443f-a65b-d21fe96a405b
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AST_EXTENSION_NOT_INUSE.
(closes issue #16035)
Reported by: francesco_r
Patches:
pbx.c.patch uploaded by viniciusfontes (license 978)
Tested by: francesco_r, agx, lawbar
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277327 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17636)
Reported by: bklang
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277261 f38db490-d61c-443f-a65b-d21fe96a405b
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and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
AST-362
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@277247 f38db490-d61c-443f-a65b-d21fe96a405b
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don't crash when it is.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@276652 f38db490-d61c-443f-a65b-d21fe96a405b
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For SIP channels configured with the progressinband option on, the ringback was
being immediately stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the generator
triggered the check in ast_write to stop the generator. The fix here should
write the frame the same as was done before the call to ast_prod was moved.
(closes issue #17372)
Reported by: tech_admin
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@275665 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17536)
Reported by: junky
Patches:
unload_vs_mod_unload.diff uploaded by junky (license 177)
Tested by: pabelanger
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@275290 f38db490-d61c-443f-a65b-d21fe96a405b
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loaded
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Also, emit a warning if a test is registered without one of these.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@275021 f38db490-d61c-443f-a65b-d21fe96a405b
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(related to issue #15250)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274359 f38db490-d61c-443f-a65b-d21fe96a405b
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A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274157 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273884 f38db490-d61c-443f-a65b-d21fe96a405b
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while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273717 f38db490-d61c-443f-a65b-d21fe96a405b
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If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273565 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes issue SWP-1652, ABE-2240)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@273057 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17076)
Reported by: stuarth
Patches:
20100324__issue17076.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272925 f38db490-d61c-443f-a65b-d21fe96a405b
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4.4.
(closes issue #17472)
Reported by: seandarcy
Patches:
config2.patch uploaded by nivan (license 1066)
Tested by: nivan
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272921 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/750/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@272878 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_dnsmgr_entry struct.
(closes issue #15827)
Reported by: DennisD
Patches:
(modified) dnsmgr_15827.patch uploaded by chappell (license 8)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@271123 f38db490-d61c-443f-a65b-d21fe96a405b
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case-insensitive matches.
Bug reported via the -dev list. See
http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@270583 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15762)
Reported by: nblasgen
Patches:
issue15672.patch uploaded by pabelanger (license 224)
Tested by: nblasgen
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269960 f38db490-d61c-443f-a65b-d21fe96a405b
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The issue here was that the frame created when adjusting for PLC had no offset
to its audio data. If this frame were translated to another format prior to
being sent out an RTP socket, all went well because the translation code would
put an appropriate offset into the frame. However, if the SLIN audio were not
translated before being sent out the RTP socket, bad things would happen.
Specifically, the ast_rtp_raw_write makes the assumption that the frame has
at least enough of an offset that it can accommodate an RTP header. This was
not the case. As such, data was being written prior to the allocation, likely
corrupting the data the memory allocator had written. Thus when the time came
to free the data, all hell broke loose. ....Well, Asterisk crashed at least.
The fix was just what one would expect. Offset the data in the frame by a reasonable
amount. The method I used is a bit odd since the data in the frame is 16 bit integers
and not bytes. I left a big ol' comment about it. This can be improved on if someone
is interested. I was more interested in getting the crash resolved.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269821 f38db490-d61c-443f-a65b-d21fe96a405b
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This eliminates the annoying <beep> on the console.
(closes issue #17477)
Reported by: jvandal
Patches:
20100610__issue17477.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269635 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. We now handle color displays properly.
(closes issue #16784)
Reported by: pabelanger
Patches:
20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
Tested by: pabelanger, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@269334 f38db490-d61c-443f-a65b-d21fe96a405b
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without any logger levels
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@268203 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16684)
Reported by: Silmaril
Patches:
patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@267009 f38db490-d61c-443f-a65b-d21fe96a405b
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Uses the VT100 method of clearing the line from the cursor position to the
end of the line: Esc-0K
(closes issue #17160)
Reported by: coolmig
Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266585 f38db490-d61c-443f-a65b-d21fe96a405b
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signal.
If you call signal() in a Solaris signal handler, instead of just resetting
the signal handler, it causes the signal to refire, because the signal is not
marked as handled prior to the signal handler being called. This effectively
causes Solaris to immediately exceed the threadstack in recursive signal
handlers and crash.
(closes issue #17000)
Reported by: rmcgilvr
Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@266142 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
Review: https://reviewboard.asterisk.org/r/639/
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r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
Merge the rest of the FullyBooted patch
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concluded.
From reviewboard
Background:
A Digium customer discovered a somewhat odd bug. The setup is that parties A
and B are bridged, and party A places party B on hold. While party B is
listening to hold music, he mashes a bunch of DTMF. Party A takes party
B off hold while this is happening, but party B continues to hear hold
music. I could reproduce this about 1 in 5 times.
The issue:
When DTMF features are enabled and a user presses keys, the channel that
the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read
from the channel during the sleep, the frame is dropped. Thus the
unhold indication is never made to the channel that was originally placed
on hold.
The fix:
Originally, I discussed with Kevin possible ways of fixing the specific
problem reported. However, we determined that the same type of problem
could happen in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional() to
defer specific frame types so they can be re-queued once the sleep has
finished. I made a common function for determining if a frame should
be deferred so that there are not two identical switch blocks to
maintain.
Review: https://reviewboard.asterisk.org/r/674/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264996 f38db490-d61c-443f-a65b-d21fe96a405b
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Several callers of ast_callerid_parse() do not initialize the name
parameter before calling thus there is the potential to use an
uninitialized pointer.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264820 f38db490-d61c-443f-a65b-d21fe96a405b
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Analogous to trunk revision 264452, but without the change
to chan_sip since it is not necessary in this branch.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@264541 f38db490-d61c-443f-a65b-d21fe96a405b
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persist states when detecting multitone sequences.
(closes issue #16749)
Reported by: dant
Patches:
dsp.c-bug16749-1.patch uploaded by dant (license 670)
Tested by: dant
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263949 f38db490-d61c-443f-a65b-d21fe96a405b
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When using strsep, if one of the list of specified separators is not found,
it is the first parameter to strsep which is now NULL, not the pointer returned
by strsep.
This issue isn't especially severe in that the worst it is likely to do is waste
some cycles when a device with no '/' and no ':' is passed to ast_device_state.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263639 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17257)
Reported by: tim_ringenbach
Patches:
hints_crash_fix.diff uploaded by tim ringenbach (license 540)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263637 f38db490-d61c-443f-a65b-d21fe96a405b
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The Version field in the cookies we're setting contain quotes around the version
number which is not compatible with RFC2109 and breaks some implementations.
(closes issue #17231)
Reported by: ecarruda
Patches:
manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559)
manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559)
Tested by: ecarruda, russell
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263456 f38db490-d61c-443f-a65b-d21fe96a405b
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dahdi_compat.h was not being included in channel.c when used with
Zaptel and wasn't in file.c at all.
(closes issue #15250)
Reported by: mneuhauser
Patches:
dahdi_compat.patch uploaded by mneuhauser (license 425)
Tested by: IgorG
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@263112 f38db490-d61c-443f-a65b-d21fe96a405b
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