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
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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 )
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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
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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/
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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
During the process of removing an audiohook from one channel
and attaching it to another the audiohook's status is updated
to DONE and then back to whatever it was previously. Typically
updating the status after setting it to DONE is not a good idea
because DONE can trigger unrecoverable audiohook destruction
events... because of this a conditional check was added to
audiohook_update_status to explicitly prevent the audiohook
from ever changing after being set to DONE. It was this check
that prevented audiohook inherit from work properly though.
Now ast_audiohook_move_by_source is treated as a special exception,
as the audiohook must be returned to its previous status after
attaching it to the new channel. This is only a safe operation
because the audiohook's lock is held the entire time, otherwise
this could cause trouble.
(closes issue #16522)
Reported by: corruptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_transfer sets res to 0 if there is no technology transfer function,
but then tests for it to be negative before deciding to do an early exit.
As a result, it will will wait for an AST_CONTROL_TRANSFER message that
will never come.
(closes issue #16424)
Reported by: davidw
Patches:
Issue_16424_trunk_234134.patch uploaded by davidw (license 780)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
fixes crash in "scheduled_destroy" in chan_iax
A signed short was used to represent a callnumber. This is makes
it possible to attempt to access the iaxs array with a negative
index.
(closes issue #16565)
Reported by: jensvb
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The channel name comparison was not comparing the whole string and therefore
if one channel name was a substring of the other, the bridge would fail.
(closes issue #16528)
Reported by: telecos82
Patches:
res_features_r236843.diff uploaded by telecos82 (license 687)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
