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When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.
This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.
Along the way, this patch also makes a few other minor changes:
* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
more closely match what is used during run-time configuration.
* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
instead of AC_PATH_PROG.
* Adds a new variable for linker flags in the build system (DYLINK), used for
producing true shared libraries (as opposed to the dynamically loadable
modules that the build system produces for 'regular' Asterisk modules).
* Moves the Makefile bits that handle installation and uninstallation of the
main Asterisk binary into main/Makefile from the top-level Makefile.
* Moves a couple of useful preprocessor macros from optional_api.h to
asterisk.h.
Review: https://reviewboard.asterisk.org/r/1006/
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Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
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Fixes up softkey endcall. Previous code was a copy of onhook, now
allows for endcall softkey to be used while device is still onhook.
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The previous change made the code look for emN and pciN in addition to what
it did originally, which was search for ethN. However, it needed to be looking
for pciN#N, so that's what it does now.
This also moves the memset() to be before every ioctl().
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Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM)
audio for quite some time, but some endpoints refer to it as 'L16-256'. This
commit adds this as an alias for the existing format.
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As of Fedora 15, ethN is not the name of ethernet interfaces. The names
are emN or pciN. Update some code that searched for interfaces named
ethN to look for the new names, as well. For more information about why
this change was made, see this page:
http://domsch.com/blog/?p=455
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Fix double format_cap iterator cleanup.
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AMI session.
I also went ahead and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's
how we handle this stuff these days.
(closes issue ASTERISK-19249)
Reporter: Jamuel Starkey
Patches:
res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766)
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Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as
a result.
Review: https://reviewboard.asterisk.org/r/1697/
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This patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show whether or
not CALENDAR_WRITE has passed. This patch also adds some debugging for caldav
PUT responses and no longer treats responses with no body as an error (as a PUT
gets a 201 Created with no body).
(closes issue ASTERISK-16903)
Reported by: Clod Patry
Tested by: Terry Wilson
Patches:
calendarstatus.diff uploaded by Clod Patry (License #5138), slightly modified by Terry Wilson
Review: https://reviewboard.asterisk.org/r/1692/
- This line, and those below, will be ignored--
M res/res_calendar.c
M res/res_calendar_exchange.c
M res/res_calendar_caldav.c
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r352863 | alecdavis | 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) | 19 lines
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r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines
rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer.
If a BLF subscription exists for long enough, using %d may print negative version numbers.
Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1694/
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sounds).
(Closes issue ASTERISK-19233)
Reported by: Matt Behrens
Patches:
chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
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For whatever reason, we don't have a single function for copying data like this
from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the
sip_pvt, but it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a peer to a
private.
(Closes issue ASTERISK-19029)
Reported by: Matt Lehner
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r352705 | alecdavis | 2012-01-26 19:33:11 +1300 (Thu, 26 Jan 2012) | 27 lines
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r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines
Cleanup dialog-info+xml Notify dialog
Make similar to other Notify messages.
sample output:
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="715" state="full" entity="sip:8523@192.168.x.xx">
<dialog id="8523">
<state>terminated</state>
</dialog>
</dialog-info>
Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1693/
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A long time ago, in a land far far away, we added "asterisk/ast_version.h",
which provides the ast_get_version() and ast_get_version_num() functions. These
were added so that modules that needed the version information for the Asterisk
instance they were loaded in could actually get it (as opposed the version that
they were compiled against). We changed everything in the tree to use the
new mechanism (although later main/test.c was added using the old method).
However, the old mechanism was never removed, and as a result, new code is
still trying to use it.
This commit removes asterisk/version.h and replaces it with a header that
will generate a compile-time error if you try to use it (the error message
tells you which header you should use instead). It also removes the Makefile
and build_tools bits that generated the file, and it updates main/test.c to
use the 'proper' method of getting the Asterisk version information.
This is an API change and thus is being committed for trunk only, but it's
a fairly minor one and definitely improves the situation for out-of-tree
modules.
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Avoid unnecessary rebuilds of main/test.c.
main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
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* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.
* Pass up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that this is a
behavior change that could concievably affect some people.)
* Block user from adding standard headers that are added automatically.
(To, From,...)
* Allow the user to override the Content-Type header contents sent by
MessageSend.
* Decrement Max-Forwards header if the user transferred it from an
incoming message.
* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.
* Documents what SIP expects in the MessageSend(from) parameter.
(closes issue ASTERISK-18992)
Reported by: Yuri
(closes issue ASTERISK-18917)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/1683/
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1) Be sure and free at unload the epa_backend we allocate at startup
2) Do the same sip_registry cleanup at unload we do at reload
Review: https://reviewboard.asterisk.org/r/1689/
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These files have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between 'modified'
and 'unmodified' states.
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There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.
(closes issue ASTERISK-15537)
Reported by: yarique
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proxy configured.
(closes issue ASTERISK-16550)
reported by: Olle Johansson
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Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!
(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
confbridge.tar.001 uploaded by Cameron Twomey
confbridge.tar.002 uploaded by Cameron Twomey
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Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
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* Fixed a potential memory leak when an existing datastore is manually
destroyed by inline code instead of calling ast_datastore_free().
(closes issue ASTERISK-17948)
Reported by: Archie Cobbs
Review: https://reviewboard.asterisk.org/r/1687/
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Move RTP timeout check to before bridged channel check so it is actually executed.
(issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury
(closes issue ASTERISK-14534)
Reported by: kriborgen
Patches:
chan_sip.patch uploaded by kriborgen (license 6138)
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This prevents the 'h' extension from being run on the transferee
channel when it is transferred via a native transfer mechanism such
as SIP REFER.
(closes ASTERISK-19173)
Reported by: Ross Beer
Tested by: Kristjan Vrban
Patches:
ASTERISK-19173 by Mark Michelson (license 5049)
Review: https://reviewboard.asterisk.org/r/1685
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While the FAXOPT function could be used to set the modem capabilities, the
input to that function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting the spandsp
layer.
(closes issue: ASTERISK-16409)
Reported by: Kristijan Vrban
Tested by: Matt Jordan, Matthew Nicholson
Patches:
spandsp-modems-1.8.diff uploaded by mnicholson (license 5081)
spandsp-modems-10.diff uploaded by mnicholson (license 5081)
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hold/between songs
This is a feature patch which allows an 'announcement' option to be specified in
musiconhold.conf which should be set to the name of a sound. If a valid sound is
specified for this option, then it will be played on that music on hold class whenever
a channel bound to that class is put on hold as well as when Asterisk is able to detect
that a song has ended before starting the next song (excludes external players).
(closes ASTERISK-18977)
Reported by: Timo Teräs
Patches:
asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409)
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monitor launch.
MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/
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The invalid value used when notifycid was enabled was benign. As far as
the code was concerned -1 and 1 are equivalent.
(closes issue ASTERISK-19232)
Reported by: Eike Kuiper
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Note: Noone calls ast_app_dtget() with the timeout parameter of zero so
the bad code normally will never get executed.
* Fix unnecessary floating point division in func_timeout.c
timeout_write() when all other values are integers.
(closes issue ASTERISK-16817)
Reported by: Dmitry Andrianov
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If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
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(closes issue ASTERISK-16877)
Reported by: klaus3000
Patches:
show-complete-routeset-patch.txt (license #5054) patch uploaded by klaus3000 (modified)
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Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.
Event description:
Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id
`source` can be either RTPTimeout or SIPSessionTimer
(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson
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* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled
(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.
Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett
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Don't be alarmed. This only affected trunk, and it would have
required manager access to your system.
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These changes are in a file that is not compiled by default, and so were
missed on earlier checks.
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These functions are not noops and modify the array that is passed in. Thanks
for the catch Richard.
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GCC 4.6.3 caught these in dev mode as well.
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(closes ASTERISK-19057)
Reported By: Yuri
Patches: 348360chan_sip.diff uploaded by Yuri (license 5242)
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gcc version 4.6.2 caught an unused variable in the ilbc codec
library. This would prevent compilation with --enable-dev-mode;
variable removed.
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* Fix corner cases in get_calleridname() parsing and ensure that the
output buffer is nul terminated.
* Make get_calleridname() truncate the name it parses if the given buffer
is too small rather than abandoning the parse and not returning anything
for the name. Adjusted get_calleridname_test() unit test to handle the
truncation change.
* Fix get_in_brackets_test() unit test to check the results of
get_in_brackets() correctly.
* Fix parse_name_andor_addr() to not return the address of a local buffer.
This function is currently not used.
* Fix potential NULL pointer dereference in sip_sendtext().
* No need to memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it is nul
terminated.
* Reply with an accurate response if get_msg_text() fails in
receive_message(). This is academic in v1.8 because get_msg_text() can
never fail.
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Change the RTCP RR and SR generation code to convert Asterisk's internal jitter
statistics to be represented in RTP timestamp units based on the rate of the
codec in use instead of in seconds.
(closes issue ASTERISK-14530)
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