We're revisiting the previous patch, albeit with a method that overcomes the
prior criticism that it was not POSIX-compliant.
(closes issue #16602)
Reported by: frawd
Patches:
20100114__issue16602.diff.txt uploaded by tilghman (license 14)
Tested by: frawd
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240499 f38db490-d61c-443f-a65b-d21fe96a405b
Asterisk would crash on startup if MALLOC_DEBUG were set in menuselect. This is because
the manager action UpdateConfig had to resize its string field allocation to set the
description. When the resize occurred, ast_copy_string would crash because we were
attempting to copy a string from a NULL pointer. Setting the strings initially makes
the code much less crashy.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240420 f38db490-d61c-443f-a65b-d21fe96a405b
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
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240415 f38db490-d61c-443f-a65b-d21fe96a405b
The problem was the OUTGOING flag was not getting set properly on the channel,
resulting in pickup failing as ast_read thought the call was inbound. Refer to
170393 for a more verbose description as this is the same exact change.
(closes issue #16539)
Reported by: syspert
Patches:
bug16539.patch uploaded by jpeeler (license 325)
Tested by: syspert
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240179 f38db490-d61c-443f-a65b-d21fe96a405b
Add a how-to set of documentation about building queues with Asterisk.
This documentation is based on Asterisk 1.6.2 but should work on most
versions with minor modifications.
(closes issue #16237)
Reported by: lmadsen
Patches:
Building Queues (FINAL).txt uploaded by lmadsen (license 10)
Tested by: pdhales, lmadsen, cmdrwalrus
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240039 f38db490-d61c-443f-a65b-d21fe96a405b
Specifically, by setting TESTTIME() to a particular date and time, you
can test whether a dialplan correctly branches as was intended. This was
developed after recent questions on the -users list on how to test their
holiday dialplan logic.
(closes issue #16464)
Reported by: tilghman
Patches:
20100112__issue16464.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/458/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239957 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010) | 11 lines
Fix regression for timed out parked call returning to caller
This issue seems to have been exposed by the fix in 160390 whereby using a
masquerade prevented a crash. The new channel used in the masquerade was
not copying the macro information from the old channel.
(closes issue #15459)
Reported by: djrodman
Patches:
patch_15459.txt uploaded by mnick (license )
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239839 f38db490-d61c-443f-a65b-d21fe96a405b
Also, made a Makefile change to ensure that the expression parser C source files get
regenerated correctly, when we need that to happen.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239797 f38db490-d61c-443f-a65b-d21fe96a405b
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r239718 | dvossel | 2010-01-13 11:16:12 -0600 (Wed, 13 Jan 2010) | 23 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 0016524)
Reported by: kobaz
(closes issue 0016523)
Reported by: kobaz
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/456/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239719 f38db490-d61c-443f-a65b-d21fe96a405b
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
(closes issue #15819)
Reported by: klaus3000
Patches:
asterisk-sip-show-channelstats-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000, oej
This patch is for trunk only and will be blocked in 1.6.2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239663 f38db490-d61c-443f-a65b-d21fe96a405b
The second is the default state for matching CID in the dialplan (no matching)
while the first matches one particular CallerID. This is a regression.
(fixes AST-314, SWP-611)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239571 f38db490-d61c-443f-a65b-d21fe96a405b
The code that handled setting 'm=text' in the sdp was not executing
in the correct order. The check to see if text was needed came after
the check to add 'm=text' to the sdp, this resulted in 'm=text' always
being set to 0 because it looked like text was never required.
(closes issue #16457)
Reported by: peterj
Patches:
textportinsdp.diff uploaded by peterj (license 951)
issue16457.diff uploaded by dvossel (license 671)
Tested by: peterj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239427 f38db490-d61c-443f-a65b-d21fe96a405b
dvossel