set to make sure that when we come back out of alarm, it gets reported in the log
and manager interface (after discussion with tzafrir on the -dev list)
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does not know what to do with these alarms. Only Asterisk 1.6 cares about it.
So, if we get an unknown alarm in chan_zap, don't generate confusing log messages
about it.
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This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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which event was received. What actually was happening was that it was reporting the number of bytes returned
from a call to read().
Thanks to Jared Smith for bringing the issue up on IRC
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in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
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could lead to a looped list, meaning it will traverse forever.
(closes issue #11818)
Reported by: michael-fig
Patches:
11818.patch uploaded by putnopvut (license 60)
Tested by: michael-fig
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it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
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or hostname was provided, which is the one part of the dial string that is
absolutely required. If it's not there, bail out.
(closes issue #11897)
Reported by sokhapkin
Patch by me
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Reported by: MVF
Patches:
bug11736-2.diff uploaded by oej (license 306)
Tested by: oej, MVF, revolution
(russellb: This was the showstopper for the release.)
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internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old
head to avoid to heavy memory allocations on some systems.
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masquerade, the monitors on the two channels involved are swapped. In 99% of the cases
this results in the desired effect. However, if monitoring a local channel, this caused
the monitor which was on the local channel to get moved onto a channel which is immediately
hung up after the masquerade has completed. By swapping the monitors prior to the masquerade,
we avoid the problem by tricking the masquerade into placing the monitor back onto the channel
where we want it.
During the investigation of the issue, the channel's monitor was the only thing that was swapped
in such a manner which did not make sense to have done. All other variable swapping made sense.
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the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
(closes issue #11730)
Reported by: UDI-Doug
Patches:
11730.patch uploaded by putnopvut (license 60)
Tested by: UDI-Doug
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buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
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output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549.
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This deadlock was introduced by the fix to ensure that channels are properly
locked when handling channel variables. There were sections of this code where
the channel pvt was locked before the channel lock, when in fact it _must_ be
the other way around.
(closes issue #11582)
Reported by: bugi
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I wonder if Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends on it.
Reported by: caio1982
(closes issue #11574)
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rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.
While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.
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We do not want to do this (see bug below for details).
This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.
Isue #10690.
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to reinvite each other in 1.4 and trunk.
This patch implements support for the 491 error code that
Asterisk 1.4 generates on situations where we get an
incoming INVITE and already has one in progress.
Thanks to mavetju for reporting and to Raj Jain for an
excellent explanation of the problem.
Patch by myself. Tested with 8 Asterisk servers connected
to each other in a training network.
Closes issue #10481
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This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
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codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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ast_uri_decode function as opposed to my home-rolled one. Also added
comments.
Thanks to oej for pointing me in the right direction
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in extensions.conf AND maintain their escaped characters when forming URI's
(closes issue #10681, reported by cahen, patched by me, code review by file)
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asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
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and ast_string_field_free_all to ast_string_field_reset_all
to avoid misuse (due to too similar names and an error in
documentation). Fix two related memory leaks in app_meetme.
No need to merge to trunk, different fix already applied there.
Not applicable to 1.2
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Terry Wilson
Jason Parker
Kevin P. Fleming
Tilghman Lesher
Russell Bryant
Joshua Colp
Mark Michelson
Olle Johansson
Christian Richter
Philippe Sultan
Luigi Rizzo