https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines
Ensure that monitor recordings are written to the correct location.
Recordings should be placed in the monitor directory when a non-absolute path
is used.
Exact dialplan used for testing:
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
exten => 5040, n, dial(sip/5001)
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
exten => 5041, n, dial(sip/5001)
exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001)
ABE-2101
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248793 f38db490-d61c-443f-a65b-d21fe96a405b
The new JabberStatus event gives a concise view of the status change to the AMI
clients. Thanks fiddur!
(closes issue #16760)
Reported by: fiddur
Patches:
244498.2.diff uploaded by fiddur (license 678)
Tested by: fiddur, phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247500 f38db490-d61c-443f-a65b-d21fe96a405b
Detect all platforms that don't like that, either, and ensure that when documentation is
missing, we pass a non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689)
Reported by: bklang
Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/497/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246030 f38db490-d61c-443f-a65b-d21fe96a405b
Initialize the calendars container before calling load_config and return FAILURE
on allocation failure. Also, use the AST_MODULE_LOAD_* values for return values.
Thanks to rmudgett for pointing out the error and the need to use the defined
values for return
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242812 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242521 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237406 f38db490-d61c-443f-a65b-d21fe96a405b
- Add dependency in chan_mgcp that was missing
- Add a small amount of doc to the source code
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237284 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines
Change Monitor to not assume file to write to does not contain pathing.
227944 changed the fname_base argument to always append the configured monitor
path. This change was necessary to properly compare files for uniqueness.
If a full path is given though, nothing needs to be appended and that is
handled correctly now.
(closes issue #16377)
(closes issue #16376)
Reported by: bcnit
Patches:
res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235941 f38db490-d61c-443f-a65b-d21fe96a405b
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
* Classes are now tracked past removal from the core container, and module
removal is actively prevented until all references are freed.
* A hanging reference stored in the channel has been removed. This could have
caused a mismatch and the music state not properly cleared, if two or more
reloads occurred between MOH being stopped and MOH being restarted.
* In certain circumstances, duplicate classes were possible.
* A race existed at reload time between a process being killed and the thread
responsible for reading from the related pipe respawning that process.
* Several reference counts have also been corrected. At least one could have
caused deleted classes to stick around forever, consuming resources. This
originally manifested as MOH external processes that were not killed at
reload time.
(closes issue #16279, closes issue #16207)
Reported by: parisioa, dcabot
Patches:
20091202__issue16279__2.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232660 f38db490-d61c-443f-a65b-d21fe96a405b
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
A thread storage variable was being freed incorrectly, which
resulted in a double free if two queries were made in the same thread.
(closes issue #16011)
Reported by: cristiandimache
Patches:
issue16011.diff uploaded by dvossel (license 671)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229093 f38db490-d61c-443f-a65b-d21fe96a405b
* 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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines
Fix incorrect filename comparsion after monitor file change
The logic to detect if a requested file is indeed a different file from the
current file was incorrect. The main issue being confusion of the use of
filename_base which was previously set without pathing information and then
compared to another full path. Robust file comparison logic has been added
to properly check if two files are the same even if symlinks are used.
(closes issue #15313)
Reported by: caspy
Patches:
20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325)
but mostly tilghman's work
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227945 f38db490-d61c-443f-a65b-d21fe96a405b
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
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines
Correct timestamp calculations when RTP sample rates over 8kHz are used.
While testing some endpoints that support 16kHz and 32kHz sample rates, some
log messages were generated due to calc_rxstamp() computing timestamps in a way
that produced odd results, so this patch sanitizes the result of the
computations.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224671 f38db490-d61c-443f-a65b-d21fe96a405b
CONFIRMED status doesn't imply busy or free, that is handled with the TRANSP
field. Luckily, libical already sets the is_busy status on the span for us.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223370 f38db490-d61c-443f-a65b-d21fe96a405b
This isn't the best way to do this, but it is the easiest. There are some
limitations that are going to need to be addressed at some point with reloads
and when I (or someone else) work on that, then the API can be updated to
handle passing the private config data that the calendar tech modules need in
a better way as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223016 f38db490-d61c-443f-a65b-d21fe96a405b
This change is done in such a way as to allow the driver to continue to
function with older databases which don't have these features.
(closes issue #16000)
Reported by: jamicque
Patches:
20091002__issue16000.diff.txt uploaded by tilghman (license 14)
20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222309 f38db490-d61c-443f-a65b-d21fe96a405b
This affected the ~~EXTEN~~ hack, where a subroutine might have changed the
value before it was used in the caller.
Patch by myself, tested by ebroad on #asterisk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222273 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222176 f38db490-d61c-443f-a65b-d21fe96a405b
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/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
gcc 4.4 has more strict rules for aliasing. It doesn't like a
struct sockaddr_in pointer pointing to a struct sockaddr. So we make it
a union.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217445 f38db490-d61c-443f-a65b-d21fe96a405b
This makes res_calendar.c compile on OpenBSD and the same
cast is used in a lot of other places where time_t type vars are used.
(closes issue #15656)
Reported by: mvanbaak
Patches:
2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak (license 7)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212343 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207680 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces the security events API. This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication. These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.
Inside of Asterisk, the events go through the ast_event API. This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.
One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level. Using logger.conf, these log entries may be
sent to a file, or to syslog.
One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip. That will be more complicated and will
be done as its own project as the next phase of security events work.
For more information on the security events framework, see the documentation
generated from doc/tex/. "make asterisk.pdf"
Review: https://reviewboard.asterisk.org/r/273/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206021 f38db490-d61c-443f-a65b-d21fe96a405b
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205120 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
Fix memory corruption and leakage related reloads of non files mode MoH classes.
For Music on Hold classes that are not files mode, meaning that we are executing
an application that will feed us audio data, we use a thread to monitor the
external application and read audio from it. This thread also makes use of the
MoH class object. In the MoH class destructor, we used pthread_cancel() to ask
the thread to exit. Unfortunately, the code did not wait to ensure that the
thread actually went away. What needed to be done is a pthread_join() to ensure
that the thread fully cleans up before we proceed. By adding this one line, we
resolve two significant problems:
1) Since the thread was never joined, it never fully goes away. So, on every
reload of non-files mode MoH, an unused thread was sticking around.
2) There was a race condition here where the application monitoring thread
could still try to access the MoH class, even though the thread executing
the MoH reload has already destroyed it.
(issue #15109)
Reported by: jvandal
(issue #15123)
Reported by: axisinternet
(issue #15195)
Reported by: amorsen
(issue AST-208)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201610 f38db490-d61c-443f-a65b-d21fe96a405b
This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200519 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
(closes issue #15191)
Reported by: alecdavis
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/262/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199743 f38db490-d61c-443f-a65b-d21fe96a405b
Move MusicOnHold, SetMusicOnHold, StartMusicOnHold, StopMusicOnHold static
documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
res_musiconhold_static_conversion.txt uploaded by lmadsen (license 10)
(with some fixes and formatting by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199413 f38db490-d61c-443f-a65b-d21fe96a405b
Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
AstXML form.
(issue #15245)
Reported by: eliel
Patches:
res_phoneprov_static_conversion.txt uploaded by lmadsen (license 10)
(with PP_EACH_USER add by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199411 f38db490-d61c-443f-a65b-d21fe96a405b
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198661 f38db490-d61c-443f-a65b-d21fe96a405b
aji_io_recv takes the maximum number of bytes to read (instead of the total
buffer size), so we have to subtract 1 from our buffer size. Without this, when
we receive packets that are larger than our buffer, iksemel will choke and
things get wonky.
(closes issue #15232)
Reported by: lp0
Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198375 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
Properly terminate AMI JabberSend response messages.
The response message (either Error or Success) needs an extra trailing \r\n
after the fields to inform the client that the message is complete.
(closes issue #14876)
Reported by: srt
Patches:
05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198371 f38db490-d61c-443f-a65b-d21fe96a405b
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198146 f38db490-d61c-443f-a65b-d21fe96a405b
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197738 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines
Resolve a file handle leak.
The frames here should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became reference counted,
this code would leak the file stream for the file being read.
(closes issue #15181)
Reported by: jkroon
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196843 f38db490-d61c-443f-a65b-d21fe96a405b
'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196758 f38db490-d61c-443f-a65b-d21fe96a405b
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
Since we are dealing with a 'const char * const' now, we have to create a
temporary copy of the string to work on rather than the original. Fix inspired
by reporter. Reviewed by everyone-and-their-mother in #asterisk-dev.
(closes issue #15184)
Reported by: andrew
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196270 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
Move the AGI commands 'receive text', 'receive char' and 'record'
static documentation to XML docs.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195365 f38db490-d61c-443f-a65b-d21fe96a405b
In res_timer_timerfd, handle the case that set_rate gets called while a timer
is still in continuous mode. In this case, we want to remember the configured
rate, but not actually set it until continuous mode has been disabled.
Thanks to dvossel for finding and helping to debug the problem.
(closes issue #15080)
Reported by: dvossel
Tested by: dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193718 f38db490-d61c-443f-a65b-d21fe96a405b
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines
Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
(closes AST-212)
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r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines
Fix a typo from 190661.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190663 f38db490-d61c-443f-a65b-d21fe96a405b
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines
Properly handle @s within hints in AEL.
AEL was not handling the case of a device hint containing an @ symbol, which
caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
This patch makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim.
(closes issue #14941)
Reported by: bpgoldsb
Patches:
bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@189464 f38db490-d61c-443f-a65b-d21fe96a405b
The code will now only change the address and port. It will not overwrite any other values.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187773 f38db490-d61c-443f-a65b-d21fe96a405b
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
(closes issue #14661)
Reported by: sum
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187421 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187046 f38db490-d61c-443f-a65b-d21fe96a405b
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
jpeeler