https://origsvn.digium.com/svn/asterisk/branches/1.4
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r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines
Modify directory name reading to be interrupted with operator or pound escape.
In the case of accidentally entering the wrong first three letters for the
reading, users could be very frustrated if the name listing is very long. This
allows interrupting the reading by pressing 0 or #. 0 will attempt to execute
a configured operator (o) extension and # will exit and proceed in the
dialplan.
ABE-2200
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines
Fix logic error when checking for a devstate provider.
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sean Bright pointed out that we lost a set of newline characters in commit
190349 on a line I had recently changed. Yay for code review on commits.
(issue #17231, #10961)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines
Manager cookies are not compatible with RFC2109.
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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) | 8 lines
Update link to new version of core sounds.
The latest version of the core sounds files 1.4.19 now includes the missing
queue-minute sound file which is called by app_queue but which has been
missing.
(closes issue #17123)
Reported by: n8ideas
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds the following two commands to live_ast:
* rsync [user]@host directory
Copy over all generated files to <directory> at remote host.
Would allow running live_ast there. Hence allows separating a build
machine from a test machine.
* gen-live-asteris: regenerate live/asterisk . Useful if copying over
files to a different directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As pointed out by 'akshayb' on #asterisk-dev, the code here called a list of
bucket entries a 'bucket', and the entries within the bucket were called
'bucket_list'. This made the code very hard to understand without reading
all of it... so I've renamed 'bucket_list' to 'bucket_entry' to clarify the
purpose of the structure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Very unfortunate things happen if we add an iax_frame
to the frame queue and let go of the lock before scheduling
the frame's transmit... There is a race condition that
exists where the frame can be removed from the frame_queue
and freed before the transmit is scheduled if we do not
hold on to that lock. This results in a freed frame
being scheduled for transmit later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop. Additionally, this adds a res_timing interface, using kqueue timers.
Review: https://reviewboard.asterisk.org/r/543/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Marquis42 suggested a better method of doing what I wanted because I ended up
removing the WARNING message for all instances when really I just wanted to
remove it for the 'return' keyword, not everything.
(issue #17145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines
fixes app_meetme dsp error
We attempted to detect silence after translating a frame
from signed linear. This caused a flooding of errors. To
resolve this the code to detect silence was moved before the
translation.
(closes issue #17133)
Reported by: jsdyer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise,
when the duplicate pseudo channel is destroyed, it frees the original
pseudo channel cc_params. The original pseudo channel is then left with a
dangling pointer for when the next duplicated pseudo channel is created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines
Forgot some conditionals around the callrerouting facility help text.
JIRA ABE-2223
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r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines
Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line)
In the case of ISDN point-to-multipoint (multidevice) you can use the
mISDN "facility calldeflect" application for call diversions from external
(PSTN) to external (PSTN). In that case this is the only way to get rid
of the two call legs to the PBX and let the calling number at the C party
become the number of the A party. In the case of ISDN point-to-point
(exchange line) the call deflection facility may not be used. Instead a
call rerouting facility has to be used.
This patch for chan_misdn.c is an extension to realize this service
(facility rerouting application). It can accept either spelling:
"callrerouting" or "callrerouteing".
The patch is tested towards Deutsche Telekom and requires a modified
version of mISDN from Digium, Inc.
Patches:
misdn_rerouteing_corrected.patch (Slightly modified.)
JIRA ABE-2223
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Clean up chan_sip.c to use new AST_CLI functions
(closes issue #17287)
Reported by: pabelanger
Patches:
issue17287.patch uploaded by pabelanger (license 224)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931
Section 5.3.2. However, this resulted in an unexpected behaviour change
to the upper layer (Asterisk).
This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2
call hangup better if the version of libpri supports it.
(issue #17104)
Reported by: shawkris
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines
Use a less silly method for modifying a flex-generated file.
The sed syntax that was used wasn't actually valid, causing some versions to
choke. This is the method that is used in 1.6.x+ for similar changes.
(closes issue #16696)
Reported by: bklang
Patches:
16696-sedfix.diff uploaded by qwell (license 4)
Tested by: qwell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As usual, the way they do things is different, so we need to account for that.
automake is versioned ala BSD/Linux, but autoconf is not. We don't actually
need to specify a version there, since AC_PREREQ will cover it for us. Things
will fail pretty loudly if AC_PREREQ isn't met.
(closes issue #16341)
Reported by: bklang
Patches:
opensolaris_bootstrap.sh uploaded by bklang (license 919)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a race condition between console_hangup()
and start_stream(). It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made. If the owner
is gone that means the channel hung up and start_stream should
be aborted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear
their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber.
This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape.
If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour.
Reported by: alecdavis
Tested by: alecdavis
Patch
vm_a_extension.diff2.txt uploaded by alecdavis (license 585)
Review: https://reviewboard.asterisk.org/r/489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads. This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.
(closes issue #17303)
Reported by: stuarth
Patches:
20100507__issue17303.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pri_dchannel thread currently violates locking order by locking the private
and then attempting to queue a frame, which needs to lock the channel. Queueing
a frame is unneccesary though and is actually a regression since sig_pri.
All the places that currently use ast_softhangup_nolock now will just set the
softhangup value directly as before.
(closes issue #17216)
Reported by: lmsteffan
Patches:
bug17216.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made more places use pri_queue_control() instead of pri_queue_frame()
and a local frame variable.
* Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no
longer releases the libpri access lock unless it is required.
* Made the pri_queue_frame() and pri_queue_control() parameter list
similar to sig_pri_lock_owner().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines
Only allow the operator key to be accepted after leaving a voicemail.
Or rather disallow the operator key from being accepted when not offered,
such as after finishing a recording from within the mailbox options menu.
ABE-2121
SWP-1267
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
