The calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be unloaded.
res_calendar can potentially create many threads and I've seen issues
where the Asterisk shutdown has failed where it looked like these
threads could be the culprit.
This patch adds unload support for res_calendar. Unloading res_calendar
will also unload the dependant tech modules as well.
(closes issue ASTERISK-16744)
Review: https://reviewboard.asterisk.org/r/1657/
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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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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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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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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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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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In order to better handle RTP sources with strictrtp enabled (which is now default in 10)
using the learning mode to figure out new sources when they change is handled by checking
for a number of consecutive (by sequence number) packets received to an rtp struct
based on a new configurable value called 'probation'. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source
has been determined.
Review: https://reviewboard.asterisk.org/r/1663/
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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
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Even though we set the frame to the ast_null_frame and return that,
the caller of the frame hook may still need the frame. This now is
a bit more careful about when it frees the frame, i.e., only under
the same conditions that applied when we duplicated it in the first
place.
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A frame that is translated via ast_translate is also duplicated via ast_frdup.
This will allocate a new frame on the heap, which needs to be free'd
at the appropriate time. This issue reporter used valgrind to find that this
occurred in res_fax's fax_gateway_framehook; a quick search through the code
showed that only place this was currently not handling the translatted frame
properly.
(closes issue ASTERISK-19133)
Reported by: Sylvain Rochet
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This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.
There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
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Prior to this patch, res_musiconhold existed at the same module priority level
as the timing sources that it depends on. This would cause a problem when
music on hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded at. This
now occurs before loading other resource modules, such that the timing source
is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474)
Reporter: Luke H
Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026)
Review: https://reviewboard.asterisk.org/r/1578/
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The function ast_srtp_protect used a common buffer for both SRTP and SRTCP
packets. Since this function can be called from multiple threads for the same
SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the
packets to become corrupted as the buffer was used by both threads
simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the problem.
(closes issue ASTERISK-18889, Reported/patch by Daniel Collins)
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The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change. However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.
* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.
* Fix ast_stun_request() return value consistency.
* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.
* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found. The stun_purge_socket() hack is no longer
required.
* Reduce ast_stun_request() error messages to debug output.
* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.
(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1595/
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As a result of the fix for ASTERISK-18039, realtime caching MOH no longer
properly resumes playing back a file between different holds in the same call.
This is because scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the filename
matched the pointer to the filename in a particular position in the array. An
easy fix is to save the filename instead of a pointer to it and then do a
strcmp instead of comparing the addresses.
(closes issue ASTERISK-18912)
Review: https://reviewboard.asterisk.org/r/1596/
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Sequence number was handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet which is basically
just a bunch of bits using an or operation. Sequence number only has 16 bits
allocated to it in an RTP packet anyway, so it would add to the next field which
just happened to be the codec. This makes sure the sequence number is set to be
a 16 bit integer regardless of architecture (hopefully) and also makes it so the
incrementing of the sequence number does bitwise or at the peak of a 16 bit number
so that the value will be set back to 0 when going beyond 65535 anyway.
(closes issue ASTERISK-18291)
Reported by: Will Schick
Review: https://reviewboard.asterisk.org/r/1542/
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The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
application because the channel needed to be masqueraded to prevent a
crash. Since the Park application now always masquerades the channel into
the parking lot, the special check is no longer needed. The fix also
resulted in AGI exec Park attempting to double park the call and not honor
the Park application parameters.
* Removed no longer necessary call to ast_masq_park_call() by AGI exec for
the Park application. (Reverts -r146923)
* Fix Park application to only return 0 or -1. The AGI exec Park was
causing broken pipe error messages because the Park application returned 1
on successful park.
(closes issue ASTERISK-18737)
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r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
Merged revisions 340970 via svnmerge from
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r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
(closes issue ASTERISK-18400)
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There is no documented reason to not add the query field to the varlist
returned by a realtime multi query, despite the config category being
set to its value. Of course, there is no documentation that the category
should be set to the value either. There is lots of no documentation
when it comes to realtime. But, other engines do not skip this field so
I am forcing this backend to follow the convention, because not doing so
is very silly.
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r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines
Merged revisions 340108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines
Load the proper XML documentation when multiple modules document the same application.
This patch adds an optional "module" attribute to the XML documentation spec
that allows the documentation processor to match apps with identical names from
different modules to their documentation. This patch also fixes a number of
bugs with the documentation processor and should make it a little more
efficient. Support for multiple languages has also been properly implemented.
ASTERISK-18130
Review: https://reviewboard.asterisk.org/r/1485/
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Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no
to enable dialplan faxdetect allowing more flexibility.
as soon as a fax tone is detected the framehook is removed.
there is a penalty involved in running this framehook on
non G711 channels as they will be transcoded.
CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice.
(Closes issue ASTERISK-18569)
Reported by: Myself
Reviewed by: Matthew Nicholson, Kevin Fleming
Review: https://reviewboard.asterisk.org/r/1116/
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r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines
Only change the capabilities on the gateway when
the session is been destroyed there is still
a race condition that ends in a segfault.
if the caps are changed the logic in res_fax_spandsp
will run T30 code not gateway code to end the session.
this has been experienced on a "slower" under spec system.
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r339298 | jrose | 2011-10-04 09:09:50 -0500 (Tue, 04 Oct 2011) | 19 lines
Merged revisions 339297 via svnmerge from
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r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines
Reverting revision 333265 due to component connection problems it introduces.
I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this
problem, but first it seems prudent to remove this rather broad attempt to fix it and
instead approach this problem either from the same angle but looking only at canceling
(or possibly rescheduling) the send when we absolutely know it will cause a segfault
or, if that can't be easily accomplished, strictly from the devstate side of things.
Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe.
(issue ASTERISK-18626)
(issue ASTERISK-18078)
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r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) | 14 lines
Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will
turn off the gateway but the framehook is not destroyed.
this problem happens when a gateway is attempted in the dialplan and
the device is not available i may want to do fax to mail in the server
it will not be allowed.
instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id
Reverts 338904
Fix some white space.
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r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) | 8 lines
Remove T38 Gateway capability when detaching framehook.
SET(FAXOPT(gateway)=no) does not remove the capability when
detaching the framehook.
small patch to fix this problem.
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Jonathan Rose
Terry Wilson
Richard Mudgett
Matthew Jordan
Kinsey Moore
Mark Michelson
Kevin P. Fleming
Matthew Nicholson
Tilghman Lesher
Stefan Schmidt
Paul Belanger
Walter Doekes
Gregory Nietsky
Olle Johansson