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r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines
If two pattern classes start with the same digit and have the same number of characters, they will compare equal.
The example given in the issue report is that of [234] and [246], which have
these characteristics, yet they are clearly not equivalent. The code still
uses these two characteristics, yet when the two scores compare equal, an
additional check will be done to compare all characters within the class to
verify equality.
(closes issue #15421)
Reported by: jsmith
Patches:
20091109__issue15421__2.diff.txt uploaded by tilghman (license 14)
Tested by: jsmith, thedavidfactor
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r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines
Update WARNING message.
Update a WARNING message to give a suggested fix when encountered.
(closes issue #16198)
Reported by: atis
Tested by: atis
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* chan_console accessed pvts after deallocation.
* cdr_mysql stored a pointer that was freed by realloc()
* The module loader did not check usecount on shutdown, which led to chan_iax2
reading a timer that was already unloaded.
* The event subsystem sometimes creates an event with no IEs. Due to a corner
condition, the code would read beyond the memory boundary.
* res_pktccops did not correctly check whether its monitor thread was started.
(closes issue #16062)
Reported by: alexanderheinz
Patches:
20091109__issue16062.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228798 f38db490-d61c-443f-a65b-d21fe96a405b
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r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines
fixes audiohook write crash occuring in chan_spy whisper mode.
After writing to the audiohook list in ast_write(), frames
were being freed incorrectly. Under certain conditions this
resulted in a double free crash.
(closes issue #16133)
Reported by: wetwired
(closes issue #16045)
Reported by: bluecrow76
Patches:
issue16045.diff uploaded by dvossel (license 671)
Tested by: bluecrow76, dvossel, habile
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines
Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE.
(closes issue #15271)
Reported by: chappell
Patches:
base64_fix.patch uploaded by chappell (license 8)
Tested by: kobaz
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r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines
Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core.
(closes issue #15560)
Reported by: jvandal
(closes issue #15709)
Reported by: covici
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines
fixes crash in astfd.c
(closes issue #15981)
Reported by: slavon
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Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
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(per Russell's review of another patch)
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This patch, originally submitted by jozza, enables custom modules to send actions to AMI
and receive messages from AMI via a hook interface. Included is a simple test module to
illustrate the interface.
(closes issue #14635)
Reported by: jozza
Review: https://reviewboard.asterisk.org/r/412/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227448 f38db490-d61c-443f-a65b-d21fe96a405b
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linkedid or uniqueid field to uniquely identify a CDR.
(closes issue #15180)
Reported by: Nick_Lewis
Patches:
cdr-sequence10.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227435 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14517)
(SWP-109)
Reported by: asgaroth
Patches:
bug_14517.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy, dougm, qwell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227372 f38db490-d61c-443f-a65b-d21fe96a405b
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This is a side project I've been poking at this week. The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together. There is a ton of stuff to write about, so this will
just continue to evolve over time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226606 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines
Manager output is not always NULL-terminated, so force a NULL at the end of the filestream.
(closes issue #15495)
Reported by: pdf
Patches:
20090916__issue15495.diff.txt uploaded by tilghman (license 14)
Tested by: pdf
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The code committed in revision 225405 was broken; instead of removing the unreference code,
the logic used to decide when to do it should have been reversed. This patch corrects the
situation, and makes reference counting work properly again.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225955 f38db490-d61c-443f-a65b-d21fe96a405b
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various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225445 f38db490-d61c-443f-a65b-d21fe96a405b
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When an object is being unlinked from its container *and* being returned to
the caller, we do not want to decrement the reference count after unlinking
it from the container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining reference to
the object, that could result in the object being destroyed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225405 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103)
Reported by: majorbloodnok
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subaddress.
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225357 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225244 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines
Revert 225169, as this doesn't account for the possibility of a list of frames.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines
Isolate the frame returned from ast_translate().
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
Isolate frames returned from a DSP instance or codec translator.
The reasoning for these changes are the same as what I wrote in the commit
message for rev 222878.
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r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
Pay attention to the return value of the manipulate function.
While this looks like an optimization, it prevents a crash from occurring
when used with certain audiohook callbacks (diagnosed with SVN trunk,
backported to 1.4 to keep the source consistent across versions).
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r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines
Add support for relaying early media in the features attended transfer option.
(closes issue #14828)
Reported by: licedey
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Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines
Don't use data outside of its scope.
The purpose of this code was to have a hangup frame put on the list of deferred
frames. However, the code that read the hangup frame was outside of the scope
of where the hangup frame was declared.
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r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines
Remove some unnecessary code.
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r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines
Signal timeouts by returning AST_CONTROL_RINGING when originating calls.
(closes issue #15104)
Reported by: nblasgen
Patches:
manager-timeout1.diff uploaded by mnicholson (license 96)
Tested by: nblasgen, mnicholson
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r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
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ABE-1998
Review: https://reviewboard.asterisk.org/r/395/
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Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222761 f38db490-d61c-443f-a65b-d21fe96a405b
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
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r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines
Ensure the result of the hash function is positive. Negative array offsets suck.
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(closes issue #15973)
Reported by: atis
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r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines
Fix a bunch of off-by-one errors
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chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
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Thanks, Josh.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221278 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
Avoid a potential NULL dereference.
(closes issue #15865)
Reported by: kobaz
Patches:
20090915__issue15865.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221201 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_cel_report_event would geet a reference to the
bridged channel. However, certain return paths, such
as if CEL was not enabled, would result in a reference
leak. All return paths now properly unref the channel.
(closes issue #15991)
Reported by: mmichelson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220995 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220920 f38db490-d61c-443f-a65b-d21fe96a405b
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Including version.h here causes this file to get recompiled after
every commit or update, which is not needed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220496 f38db490-d61c-443f-a65b-d21fe96a405b
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