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r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines
Merged revisions 183700 via svnmerge from
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r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
Fix a memory leak in res_monitor.c
The only way that this leak would occur is if Monitor were started
using the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
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r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines
Merged revisions 183386 via svnmerge from
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines
Fix a memory leak associated with queues.
For every attempt that app_queue made to place an outbound call to a queue member,
we would allocate a queue_end_bridge structure. When the bridge for the call had
completed, we would free the structure. Unfortunately not all call attempts actually
end up bridged to a member, so we need to be more selective of when to allocate
the structure. With this change, the allocation occurs in an area where we can
guarantee that the call will be bridged.
(closes issue #14680)
Reported by: caspy
Patches:
14680.patch uploaded by mmichelson (license 60)
Tested by: caspy
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r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines
Merged revisions 183126 via svnmerge from
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r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
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r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines
Merged revisions 183115 via svnmerge from
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r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
A user was having an issue where if an outgoing SIP call was canceled, the SIP device
would remain in use if we had not received any response to the initial INVITE we sent out.
The SIP device would remain in use until the autocongestion timer was exhausted.
I tracked down the cause of this to be the section of code I am removing here. I asked several
people what the purpose of this code was meant to be, but no one could give me any sort of
answer as to why this was here. The person who was having this issue has been using this patch
for several months and it has stopped the problems they have had.
AST-196
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r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
Improve our triggering of a T38 switchover internally when triggered by a received reinvite.
Previously we reached across the channel bridge to get the other party's SIP dialog
structure in order to trigger an outgoing reinvite. This is extremely dangerous to do
and only works if bridged to another SIP channel. This patch changes this to use the
T38 control frame method of requesting a switchover. This change also causes the SIP
channel driver to propogate back whether the switchover worked or not instead of blindly
accepting the incoming T38 reinvite.
Review: http://reviewboard.digium.com/r/200/
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r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines
Fix an issue where a T38 control frame would get dropped.
If two channels were bridged together using a generic bridge the T38
control frame would get passed up instead of being indicated on the
other channel.
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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
Allow H.323 Plus library to be used in addition to the OpenH323 library
Chan_h323 can now be compiled against both the previously supported versions of
OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
script has been modified to look in the default install location of h323 to
hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
Also, the CLI command "h323 show version" has been added which indicates which
version of h323 is in use.
(closes issue #11261)
Reported by: vhatz
Patches:
asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
Tweak the handling of the frame list inside of ast_answer().
This does not change any behavior, but moves the frames from the local frame
list back to the channel read queue using an O(n) algorithm instead of O(n^2).
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r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
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r182521 | seanbright | 2009-03-17 10:24:53 -0400 (Tue, 17 Mar 2009) | 3 lines
Don't include a space before the optional extra text that may follow a help
string.
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r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines
Merged revisions 182449 via svnmerge from
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r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines
Fix race in astdb
The underlying db1 implementation does not fully isolate the pages retrieved
from astdb, so the lock protecting accesses needs to be extended until the
copy from the shared memory structure is done.
(closes issue #14682)
Reported by: makoto
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r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines
OPENR2 uses an incorrect string value if the extension delimiter is not present.
* Fixed OPENR2 using an incorrect string value if the extension
delimiter is not present in the Dial() function. This was fixed for
SS7 and PRI in trunk -r172400.
* Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others.
* Removed trailing whitespace that appeared with OPENR2.
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r182355 | russell | 2009-03-16 15:35:58 -0500 (Mon, 16 Mar 2009) | 29 lines
Add MFC/R2 support for chan_dahdi.
This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
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r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines
Merged revisions 182281 via svnmerge from
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r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
Randomize IAX2 encryption padding
The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
Review: http://reviewboard.digium.com/r/193/
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r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines
Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution.
Previously, FILE() returned one less character than specified, due to the
terminating NULL. Both the offset and length parameters now behave
identically to the way variable substitution offsets and lengths also work.
(closes issue #14670)
Reported by: BMC
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r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines
Fix a memory leak in the ast_answer / __ast_answer API call.
For a channel that is not yet answered this API call will wait
until a voice frame is received on the channel before returning.
It does this by waiting for frames on the channel and reading them
in. The frames read in were not freed when they should have been.
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r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines
Merged revisions 181990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines
Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF.
Dynamic features defined in the applicationmap section of features.conf allow
one to specify whether the caller, callee, or both have the ability to use the
feature. The documentation in the features.conf.sample file could be interpreted
to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the
calling channel in order to allow for the callee to be able to use the features
which he should have permission to use. However, the DYNAMIC_FEATURES variable
would only be read from the channel of the participant that pressed the DTMF
sequence to activate the feature. The result of this was that the callee was
unable to use dynamic features unless the dialplan writer had taken measures
to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel.
This commit changes the behavior of ast_feature_interpret to concatenate the
values of DYNAMIC_FEATURES from both parties involved in the bridge. The features
themselves determine who has permission to use them, so there is no reason to believe
that one side of the bridge could gain the ability to perform an action that they
should not have the ability to perform.
Kevin Fleming pointed out on the asterisk-users list that the typical way that this
was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel
so that the value would be inherited by the called channel. While this works, the
documentation alone is not enough to figure out why this is necessary for the callee
to be able to use dynamic features. In this particular case, changing the code to match
the documentation is safe, easy, and will generally make things easier for people for
future installations.
This bug was originally reported on the asterisk-users list by David Ruggles.
(closes issue #14657)
Reported by: mmichelson
Patches:
14657.patch uploaded by mmichelson (license 60)
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r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines
Fix an issue with requesting a T38 reinvite before the call is answered.
The code responsible for sending the T38 reinvite did not check if an INVITE was
already being handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current INVITE is done being
handled.
(issue AST-191)
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r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines
Merged revisions 181768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
If we receive an INVITE from an endpoint and then later receive a BYE from that
same endpoint before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487.
There was logic in the code prior to this commit which seemed to exist solely to
handle this situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
channel. This made no sense since we created the owner channel when we received
the INVITE, meaning that the majority of the time we would never send the 487.
The 487 being sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial INVITE transaction.
With this commit, that logic is now correctly in place.
(closes issue #14149)
Reported by: legranjl
Patches:
14149.patch uploaded by mmichelson (license 60)
Tested by: legranjl
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r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines
Adjust translation table column widths based upon the translation times.
Previously, only 5 columns were displayed, and if a translation time exceeded
99,999 useconds, it would be displayed as 0, instead of its actual time.
(closes issue #14532)
Reported by: pj
Patches:
20090311__bug14532.diff.txt uploaded by tilghman (license 14)
Tested by: pj
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r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines
Merged revisions 181659-181660 via svnmerge from
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r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
Fix another scenario where depending on configuration the stream would not get read.
For custom commands we don't know whether the audio is coming from a stream or not
so we are going to have to read the data despite no channels.
(closes issue #14416)
Reported by: caspy
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r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
Fix logic flaw in previous commit.
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r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines
Merged revisions 181655 via svnmerge from
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r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
Fix issue with streaming MOH failing if nobody is listening.
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
(closes issue #14416)
Reported by: caspy
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r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines
Merged revisions 181423 via svnmerge from
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r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines
Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without the channel
locked. This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.
(closes issue #14623)
Reported by: guillecabeza
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r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines
Merged revisions 181340 via svnmerge from
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r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
(closes issue #14607)
Reported by: stevenla
Tested by: dvossel
Review: http://reviewboard.digium.com/r/192/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
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r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines
Merged revisions 181328 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
Fix issue where an attended transfer could not be completed under a rare scenario.
When completing an attended transfer chan_sip does a check to make sure the extension
in the URI portion of the Refer-To header is a local valid extension. We don't actually
need to check this since we know for sure the other channel is already up and talking to
the extension. Some devices do not put the extension in the Refer-To header either, which
can cause the extension check to fail. We now no longer do this check if it is an attended
transfer.
(closes issue #14628)
Reported by: sverre
Patches:
14628.diff uploaded by file (license 11)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
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r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines
Merged revisions 181295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
When dtmfmode was set to auto the inband DTMF detector was not setup
on outgoing SIP calls. This caused inband DTMF detection to fail.
The inband DTMF detector is now setup for both dtmfmode inband and auto.
(closes issue #13713)
Reported by: makoto
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
