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r240842 | dvossel | 2010-01-18 09:52:55 -0600 (Mon, 18 Jan 2010) | 2 lines
fixes spelling error. s/memeber/member
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r240415 | tilghman | 2010-01-15 14:54:24 -0600 (Fri, 15 Jan 2010) | 22 lines
Merged revisions 240414 via svnmerge from
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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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r239712 | dvossel | 2010-01-13 10:31:14 -0600 (Wed, 13 Jan 2010) | 24 lines
add silence gen to wait apps
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/
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r238361 | dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
cli 'queue show' formatting fix. queue name was truncated over 12 characters
(closes issue #16078)
Reported by: RoadKill
Patches:
quequename_limit.patch uploaded by ppyy (license 906)
Tested by: dvossel
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r238181 | jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
Fix misreverting from 177158.
(closes issue #15725)
Reported by: shanermn
Patches:
v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn
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r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) | 14 lines
Merged revisions 238009 via svnmerge from
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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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This was previously committed to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and missed). 1.6.2 also needs this patch to resolve the bug.
(closes issue #16550)
Reported by: opticron
Patches:
apptest.diff uploaded by opticron (license 267)
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r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
fixes holdtime playback issue in app_queue
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
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r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) | 6 lines
Make CLI command 'mixmonitor start|stop <channel> work again.
(closes issue #16534)
Reported by: jlaguilar
Fix as suggested by jlaguilar in the bugreport
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r237327 | dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
app_queue segfaults if realtime field uniqueid is NULL
(closes issue #16385)
Reported by: haakon
Patches:
app_queue.c.patch uploaded by haakon (license 880)
app_queue.c.patch_v2 uploaded by dvossel (license 671)
Tested by: haakon
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r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) | 4 lines
Use recommended option, not deprecated option.
(closes issue #16515)
Reported by: ManChicken
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r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines
Merged revisions 236509 via svnmerge from
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r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
Avoid a crash with large numbers of MeetMe conferences.
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
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r236300 | tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
AGI may be invoked from outside the dialplan
(closes issue #16510)
Reported by: atis
Patches:
20091223__issue16510.diff.txt uploaded by tilghman (license 14)
Tested by: atis
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r235010 | kpfleming | 2009-12-15 08:35:46 -0600 (Tue, 15 Dec 2009) | 5 lines
spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)
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r234893 | alecdavis | 2009-12-15 15:29:50 +1300 (Tue, 15 Dec 2009) | 9 lines
fixes escape to extensions 'o' and 'a', for digits '0' and '*'
(closes issue #16437)
Reported by: alecdavis
Tested by: alecdavis
Patch
extension_o_a_fix.diff.txt uploaded by alecdavis (license 585)
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r234855 | alecdavis | 2009-12-15 13:54:44 +1300 (Tue, 15 Dec 2009) | 9 lines
ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.
(closes issue #16409)
Reported by: alecdavis
Tested by: alecdavis
Patch
bug_16409.diff.txt uploaded by alecdavis (license 585)
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r234380 | jpeeler | 2009-12-11 17:17:09 -0600 (Fri, 11 Dec 2009) | 18 lines
Merged revisions 234379 via svnmerge from
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r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
Fix talking detection status after conference user is muted.
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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r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines
Merged revisions 233116 via svnmerge from
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r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines
document and rename strip_control() in app_voicemail
(closes issue #16291)
Reported by: wdoekes
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r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines
Merged revisions 232820 via svnmerge from
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r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines
Deprecate "cz" in favor of "cs".
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)
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r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines
Prevent double closing of FDs by EIVR
This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications.
EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance
the second close would then close the FD now in use by AGI.
(closes issue #16305)
Reported by: diLLec
Tested by: thedavidfactor, diLLec
Review: https://reviewboard.asterisk.org/r/436/
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r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines
Merged revisions 232355 via svnmerge from
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r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines
Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239)
Reported by: CGMChris
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r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines
Merged revisions 231614 via svnmerge from
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents 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/
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r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
app_queue crashes randomly, often during call-transfers
This patch adds a ref to the queue_ent object's parent call_queue
in queue_exec() so the call_queue won't be destroyed
while the the queue_ent still holds a pointer to it.
(closes issue 0015686)
Tested by: dvossel, aragon
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r231134 | tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
Found a few places where queue refcounts were counted incorrectly. Also add debug statements.
(closes issue #15982, closes issue #15984)
Reported by: atis
Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis
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r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines
Merged revisions 230508 via svnmerge from
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r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines
fixes MixMonitor thread not exiting when StopMixMonitor is used
(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/
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r230381 | kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line
Fix another buglet in T.38 session teardown at the end of FAX sessions.
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r230343 | kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 lines
Ensure that only one end of a T.38 session initiates teardown at completion.
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r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | 13 lines
Merged revisions 229965 via svnmerge from
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r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 lines
Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide
a workaround for it that does not change existing behavior.
(closes issue #14426)
Reported by: macli
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r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines
Flags not initialized in app_softhangup.c, causing undefined behavior
Trivial patch [kobaz] to initialize an ast_flags = {0}
(closes issue #16129)
Reported by: kobaz
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r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
When GOSUB is invoked within an AGI, it may not exit correctly.
(closes issue #16216)
Reported by: atis
Patches:
20091110__atis_work.diff.txt uploaded by tilghman (license 14)
Tested by: atis
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r228196 | tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
Yet another error message in the dialplan (thanks, rmudgett/russellb)
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r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.
(closes issue #16167)
Reported by: marhbere
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r228191 | tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
MEETME_INFO should not return a literal error message to the dialplan.
(closes issue #15450)
Reported by: JimVanM
Patches:
meetmeinfopatch.diff.txt uploaded by dbrooks (license 790)
Tested by: JimVanM
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r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) | 4 lines
Don't crash if no arguments are passed.
(closes issue #16119)
Reported by: thedavidfactor
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r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines
Merged revisions 227827 via svnmerge from
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r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines
This patch modifies the Dial application to monitor the calling channel for 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/
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r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines
Change warning message to debug message.
app_controlplayback outputs a warning, when in fact it is normal.
(closes issue #16071)
Reported by: atis
Patches:
controlplayback_warning.patch uploaded by atis (license 242)
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r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines
Merged revisions 226889 via svnmerge from
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r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
Fix a bug where the recorded privacy introduction file would not get removed 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)
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This backport resolves some issues handling audio frames during FAX processing,
and ensures that the FAX application doesn't accidentally get notified of a T.38
switchover at the end of a successful FAX.
(closes issue #16127)
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r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
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r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines
Merged revisions 225105 via svnmerge from
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r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103)
Reported by: majorbloodnok
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r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines
Merged revisions 224565 via svnmerge from
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r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines
Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it.
(closes issue #14763)
Reported by: cupotka
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r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines
Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's.
This corrects an issue reported on the -users list.
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r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
Readd removed ability to allow listening to one side of the call in app_chanspy
(Option o)
(closes issue #15675)
Reported by: john8675309
Patches:
issue15675patchtrunk.txt uploaded by dbrooks (license 790)
Tested by: jgutierrez on users list:
http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
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r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines
Merged revisions 223804 via svnmerge from
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r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines
Ensure ringing continues for branched calls after progress is received
While waiting for an answer, don't send progress for branched calls
for which ringing was sent.
(closes issue #15028)
Reported by: fnordian
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r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.
(closes issue #16025)
Reported by: jamicque
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r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines
Initiate T.38 switchover when acting as called party, regardless of FAX direction.
SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
to initiate a switchover to T.38, not the direction that the FAX transfer will
take place.
(closes issue #16039)
Reported by: jamicque
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r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines
Recorded merge of revisions 223213 via svnmerge from
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r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines
Fix potential memory leak in app_dial.c
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r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines
Recorded merge of revisions 222152 via svnmerge from
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
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/
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r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
Prevents from division by zero
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