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: https://origsvn.digium.com/svn/asterisk/trunk@240499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@240420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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: https://origsvn.digium.com/svn/asterisk/trunk@240415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@240179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@240039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@239957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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: https://origsvn.digium.com/svn/asterisk/trunk@239839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@239797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
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: https://origsvn.digium.com/svn/asterisk/trunk@239719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@239712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(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: https://origsvn.digium.com/svn/asterisk/trunk@239663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@239571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@239427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is an issue which only affects trunk and the new ao2_callback OBJ_MULTIPLE
implementation. When both OBJ_MULTIPLE and OBJ_NODATA are passed, only the first
object is visited, regardless of what is returned by the specified callback. This
causes a problem when we are clearing a container, i.e.:
ao2_callback(container, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL);
Only unlinks the first object. This patch resolves this.
(closes issue #16564)
Reported by: pj
Patches:
issue16564_20100111.diff uploaded by seanbright (license 71)
Tested by: pj, seanbright
Review: https://reviewboard.asterisk.org/r/457/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes a crash on Solaris.
(closes issue #16572)
Reported by: crjw
Patches:
frame_changes.patch uploaded by crjw (license 963)
Plus several others found and fixed by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when we decode received q931 packet we must do callbacks and
when we print sended q931 packet we must not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
