* 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: https://origsvn.digium.com/svn/asterisk/trunk@232660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@229093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@228798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@227945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@223370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@223016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@222309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@222273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@217445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@212343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: https://origsvn.digium.com/svn/asterisk/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@199413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@199411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@198375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: https://origsvn.digium.com/svn/asterisk/trunk@198371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@198146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'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: https://origsvn.digium.com/svn/asterisk/trunk@196758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@196270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@193718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
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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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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: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AGI dialplan applications did not destroy the speech structure automatically
if it was not destroyed by the running AGI script. They will now do this.
(issue LUMENVOX-15)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
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r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
Fix a memory leak in res_monitor.c
The only way that this leak would occur is if Monitor were started
using the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar 2009) | 5 lines
Improve the build system to *properly* remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix.
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
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r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines
Fix another scenario where depending on configuration the stream would not get read.
For custom commands we don't know whether the audio is coming from a stream or not
so we are going to have to read the data despite no channels.
(closes issue #14416)
Reported by: caspy
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r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines
Fix logic flaw in previous commit.
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r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines
Fix issue with streaming MOH failing if nobody is listening.
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
(closes issue #14416)
Reported by: caspy
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This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.
I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.
Big thanks to all who contributed comments on this.
(closes issue #14490)
Reported by: mmichelson
Patches:
timing.txt uploaded by mmichelson (license 60)
Review: http://reviewboard.digium.com/r/164/
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r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines
This patch fixes a regression of sorts that was introduced in
rev 24425.
It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only handle
about 4893 of the 6000 extens in the single context.
This was due to the regression I mentioned. To get rid of
shift/reduce conflicts, Luigi set up right-recursive lists
for globals, context elements, switch lists, and statements.
Right recursive lists got rid of the warnings, but instead, they
use up a tremendous amount of stack space when the lists are long.
I saw this a few years back, and resolved not to fix it until
someone complained. That day has arrived!
After the changes were made, I ran the regression test suite,
and there were no problems.
I took the test case the user provided, and added 100,000
extensions to the single context, that already had 6,000 extens
in it. (I'll see your 6, and raise you 100!) It takes a few minutes
to read it all in, check it and generate code for it, but no
problems.
So, I think I can say that fundamentally, there are no longer
any limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints.
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1) Add module use count handling so that timing modules can be unloaded.
2) Implement unload_module() functions for the timing interface modules.
3) Allow multiple timing modules to be loaded, and use the one with the
highest priority value.
4) Report which timing module is being use in the "timing test" CLI command.
(closes issue #14489)
Reported by: russell
Review: http://reviewboard.digium.com/r/162/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines
Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
(closes issue #14407)
Reported by: mostyn
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along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This sequence of events posed a problem
timerfd_timer_open
timerfd_timer_enable_continuous
timerfd_timer_set_rate
timerfd_timer_disable_continuous
The reason was that the timing module was written under the assumption
that timerfd_timer_set_rate would not be called between enabling and
disabling continuous mode. What happened in this situation was that
timerfd_timer_enable_continuous saved off our previously set timer (in this
situation a 0 timer, meaning it never runs out). Then timerfd_timer_disable_continuous
would restore this 0 timer, even though it logically should set the timer to be whatever
was set in timerfd_timer_set_rate.
Now the behavior in timerfd_timer_set_rate is to overwrite the saved timer that may
or may not have been set in timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called, this will not harm the
operation.
Thanks to Terry Wilson for discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
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r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines
This patch fixes a problem where a goto (or jump, in this case)
fails a consistency check because it can't find a matching
extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern
into a regex and uses the regex code to determine the match.
I tested using the AEL code the user supplied, and now,
the consistency check passes.
(closes issue #14141)
Reported by: dimas
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Prior to this patch, the value of AGISIGUP was not always
honored when set on a channel.
(closes issue #13711)
Reported by: fmueller
Patches:
13711.patch uploaded by putnopvut (license 60)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage
was for the AGI command, "asyncagi break". This patch removes this feature.
Normally, a feature would not be removed like this. However, this code is
broken and usage of it will result in a memory leak.
Usage of this feature will make the AGI code return a result of
AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed
ownership of the channel. The channel thread will exit without destroying the
channel. Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here:
1) The only way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old ast_channel
structure will be lost.
2) Until the channel redirect happens, there is no code servicing the channel.
That means nothing is reading audio or handling events coming from the
channel. This is very bad.
The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code. That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The variable "class" was being set NULL just prior to
being dereferenced in an ao2_link call. I have moved
the setting of the variable to NULL until after the
ao2_link call.
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r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines
Set the process group ID on the MOH process so that all children will get killed
(closes issue #14099)
Reported by: caspy
Patches:
res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645)
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r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines
Fix some memory leaks found while looking at how realtime
configs are handled.
Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing
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is used while continuous mode was already turned on.
(closes issue #13738)
Reported by: smurfix
Patches:
res.patch.fixed uploaded by smurfix (license 547)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines
(closes issue #13229)
Reported by: clegall_proformatique
Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams.
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line
In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
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r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines
(closes issue #14019)
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
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because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
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They removed the LDAP_DEPRECATED define from their source and since we are using a couple
of deprecated function calls we should define it with a CFLAG.
Tested by me on OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps compiling.
It shouldn't break, we only define the LDAP_DEPRECATED with this which is what
all 2.2.X and older versions of OpenLDAP did in their own tree.
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it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
........
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This provides a new timing interface. In order to use it,
you must be running a Linux with a kernel version of
2.6.25 or newer and glibc 2.8 or newer.
This timing interface is a good alternative if a timing
source is necessary (e.g. for IAX trunking) but DAHDI is
otherwise unnecessary for the system.
For now, this commit contains the actual work done in the
res_timing_timerfd branch. There are no notices in the README
or CHANGES files yet, but they will be added in my next commit.
The timing API of Asterisk also needs to have a bit of work done
with regards to choosing which timing interface to use. This commit
makes the choice a build-time decision, by only allowing one of
the timer interfaces to be chosen in menuselect. It would be preferable
if the choice could be made at run-time, however. The preferred timing
interface could be loaded and tested, and if it does not work, choice
number two may be used instead. That sort of thing. That is beyond
the scope of work in this branch though.
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also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
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is unloaded and then Asterisk is stopped. The problem was that
we are not unregistering the ast_moh_destroy function at exit.
(closes issue #13761)
Reported by: eliel
Patches:
res_musiconhold.c.patch uploaded by eliel (license 64)
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
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A new <agi> element is used to describe the XML documentation.
We have the usual synopsis,syntax,description and seealso for AGI commands.
The CLI 'agi show commands' command was changed to show all the documentation se
ctions.
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ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
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along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named
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'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier. All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.
This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.
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app_voicemail. Instead, include it where it is needed. This turned out to be a
relatively minor issue because other headers include logger.h as well.
Need to test -addons before merging this back to 1.6.0.
(closes issue #13605)
Reported by: tomo1657
Patches:
13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson
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This allows for the ODBC parts to work on OpenBSD as well.
99.99% of the work is done by seanbright (bow, bow) and I actually
did nothing but test and yell at him that it still didn't work :)
Thanks for helping out !
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
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when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines
(closes issue #13357)
Reported by: pj
Tested by: murf
(closes issue #13416)
Reported by: yarns
Tested by: murf
If you find this message overly verbose, relax, it's probably
not meant for you. This message is meant for probably only
two people in the whole world: me, or the poor schnook that
has to maintain this code because I'm either dead or unavailable
at the moment.
This fix solves two reports, both having to do with embedding
a function call in a ${} construct. It was tricky because the
funccall syntax has parenthesis () in it. And up till now,
the 'word' token in the flex stuff didn't allow that, because
it would tend to steal the LP and RP tokens. To be truthful,
the "word" token was the trickiest, most unstable thing in
the whole lexer. I was lucky it made this long without complaints.
I had to choose every character in the pattern with extreme
care, and I knew that someday I'd have to revisit it. Well,
the day has come.
So, my brilliant idea (and I'm being modest), was to use the
surrounding ${} construct to make a state machine and capture
everything in it, no matter what it contains. But, I have to now
treat the word token like I did with comments, in that I turn
the whole thing into a state-machine sort of spec, with new
contexts "curlystate", "wordstate", and "brackstate".
Wait a minute, "brackstate"? Yes, well, it didn't take very many
regression tests to point out if I do this for ${} constructs,
I also have to do it with the $[] constructs, too.
I had to create a separate pcbstack2 and pcbstack3 because
these constructs can occur inside macro argument lists, and
when we have two state machines operating on the same structures
we'd get problems otherwise. I guess I could have stopped at
pcbstack2 and had the brackstate stuff share it, but it doesn't
hurt to be safe. So, the pcbpush and pcbpop routines also now
have versions for "2" and "3".
I had to add the {KEYWORD} construct to the initial pattern for
"word", because previously word would match stuff like "default7",
because it was a longer match than the keyword "default". But,
not any more, because the word pattern only matches only one or
two characters now, and it will always lose. So, I made it the
winner again by making an optional match on any of the keywords
before it's normal pattern.
I added another regression test to make sure we don't
lose this in future edits, and had to fix just one regression,
where it no longer reports a 'cascaded' error, which I guess
is a plus.
I've given some thought as to whether to apply these fixes to
1.4 and the 1.6.x releases, vs trunk; I decided to put it in
1.4 because one of the bug reports was against 1.4; and it
is unexpected that AEL cannot handle this situation. It actually
reduced the amount of useless "cascade" error messages that
appeared in the regressions (by one line, ehhem). There is
a possible side-effect in that it does now do more careful
checking of what's in those ${} constructs, as far as matching
parens, and brackets are concerned. Some users may find a an
insidious problem and correct it this way. This should be
exceedingly rare, I hope.
........
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines
(closes issue #11979)
Fixes multiple parking problems:
Crash when executing a park on an extension dialed by AGI due to not returning the proper return code.
Crash when using a builtin feature that was a subset of a enabled dynamic feature.
Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked.
........
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of changes to the Set and MSet apps, so people aren't
so shocked and surprised when they upgrade from
1.4 to 1.6.
Also, for the sake of those upgrading from 1.4 to
1.6 with AEL, I provide automatic support for the
"old" way of using Set(), that still does the
exact same old thing with quotes and backslashes
and so on as 1.4 did, by having AEL compile in the
use of MSet() instead of Set(), everywhere it inserts
this code.
But, if the app_set var is set to 1.6 or higher,
it uses the "new", non-evaluative Set().
This only usually happens if the user manually
inserts this into the asterisk.conf file, or runs
the "make samples" command.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
........
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to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if
they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x
installation where a "make samples" was executed, or where they hand-edited the
asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher).
(this commit does not totally solve 13249, at least not yet)
The change involves issueing a single warning while the AEL file is loading, if:
1. app_set is present in the config file, and set to 1.6 or higher.
2. there are double quotes in an assignment statement (eg x = "hi there";)
3. the warning was not already issued.
The standalone app, aelparse, does not (yet) issue this warning. I'd have to
have it read in the asterisk.conf file, and that's a bit of hassle. I'll add
it if users request it, tho.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138815 65c4cc65-6c06-0410-ace0-fbb531ad65f3
an OOB message that answers the question of, if AGI no longer shuts down the
connection on hangup, how will FastAGI know when to stop processing the call?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines
(closes issue #13236)
Reported by: korihor
Wow, this one was a challenge!
I regrouped and ran a new strategy for
setting the ~~MACRO~~ value; I set it once
per extension, up near the top. It is only
set if there is a switch in the extension.
So, I had to put in a chunk of code to detect
a switch in the pval tree.
I moved the code to insert the set of ~~exten~~
up to the beginning of the gen_prios routine,
instead of down in the switch code.
I learned that I have to push the detection
of the switches down into the code, so everywhere
I create a new exten in gen_prios, I make sure
to pass onto it the values of the mother_exten
first, and the exten next.
I had to add a couple fields to the exten
struct to accomplish this, in the ael_structs.h
file. The checked field makes it so we don't
repeat the switch search if it's been done.
I also updated the regressions.
........
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- Changing debug messages from VERBOSE to DEBUG channel
- Adding a few todo's
- Adding a few more "XMPP"'s to compliment Jabber...
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1) If a function returns SQLITE_LOCKED, no recovery is possible.
2) An error message can be allocated, even when no error is signalled.
(closes issue #13109)
Reported by: gknispel_proformatique
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines
(closes issue #13090)
Reported by: murf
The problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that
the checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion.
A small fix to traverse into the statements of the context instead
of the context solves this issue.
I also added some commented out printfs for debug, which were pretty
handy in the face of a dorky gdb.
This was a problem around since the package was first written;
but evidently pretty rare in turning up in the field.
........
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released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev,
discovered by bbryant's change to the lock tracking code to yell at you
if a thread exits with a lock still held)
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To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed
a locally modified asterisk.tex, with changes done by this script.
(closes issue #13062)
Reported by: mvanbaak
Patches:
sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak
Feedback from Corydon. Thanks for taking the time to go through this.
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
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- change ast_settimeout() to honor max rate in edge cases of file playback
(this will make some warning messages go away at the end of playing back
a file)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
........
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the corresponding roster item has a subscription value set to "none"
or "from".
Make the code more readable.
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They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
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does not require DAHDI. It's called "pthread" because it uses a pthread
API call in the timing thread for sleeping and ensuring we wake up at
an appropriate time. I wasn't sure what else to call it. :)
The timing API requires a file descriptor that can be polled on. So,
when you open a timer, this module creates a pipe and returns the read
end of the pipe. There is a background thread that wakes up every 10ms
and checks to see if any of the currently open timers need a 'tick' and
writes to the appropriate pipe.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
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has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
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and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines
as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines
Switch to using ast_random() rather than just rand().
This does not fix the bug reported, but I believe it is correct.
(from issue #12446)
Patches:
bug_12446.diff uploaded by snuffy (license 35)
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This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) | 8 lines
Add special case for when the agi cannot be executed, to comply with the documentation that
we return failure in that case.
(closes issue #12462)
Reported by: fmueller
Patches:
20080416__bug12462.diff.txt uploaded by Corydon76 (license 14)
Tested by: fmueller
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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines
(closes issue #12302)
Reported by: pj
Tested by: murf
These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.
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G.722 music on hold working for me.
(issue #12164, reported by milazzo and jsmith, patches by me)
res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
ast_write(). The reason that this was a bug is that the G.722 frame
originally says it has 320 samples in it (which is correct). However,
when the frame is written to a channel that uses RTP, main/rtp.c modifies
the frame to cut the number of samples in half before it sends it on
the wire. This is to account for the stupid incorrect G.722 spec that
makes it so we have to lie about the number of samples with RTP. I should
probably go and re-work the RTP code so it doesn't modify the frame so
that a bug like this won't happen in the future. However, this change to
MOH is harmless.
main/channel.c:
- I made two fixes in regards to generator timing. Generators use samples
for timing. However, this code assumed 8 kHz samples. In one case, it was
a hard coded 160 samples, that is now written as the sample rate / 50. The
other place was dealing with timing a generator based on frames coming from
the other direction. However, that would have only worked if the sample
rates for the formats in both directions were the same. The code now takes
into account that the sample rates may differ, and scales the generator
samples accordingly.
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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines
Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky
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actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines
(closes issue #11903)
Reported by: atis
Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.
Sorry for not solving this earlier. I got distracted.
Added the ntest23 regression test, which is mainly a copy of ntest22,
but with a few juicy errors thrown in, to replicate the kind of
error that atis spotted.
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caching would not work properly for users.conf and any other file read from
more than one place. I needed to add the filename which requested the config
file to get it to work properly.
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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
given our removal of deadagi as an actual application.
(closes issue #12161)
Reported by: explidous
Patches:
res_agi_12161.patch uploaded by juggie (license 24)
Tested by: juggie
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan 2008) | 10 lines
1. Prevent the addition of an extra '/' to the beginning of an absolute pathname.
2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the
filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously
delete them.
(closes issue #11741)
Reported by: garlew
Tested by: putnopvut
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r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver.
(closes issue #11798)
Reported by: Corydon76
Patches:
20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak
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option is incorrectly passed to the transferee when built-in
attended transfers are used. There is still a problem with 'T',
but better to fix some problems than no problems while we work
on it.
(closes issue #7904)
Reported by: k-egg
Patches:
transfer-fix-trunk-r97657.diff uploaded by sergee (license 138)
Tested by: sergee, otherwiseguy
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the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines
If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else.
(closes issue #11743)
Reported by: davevg
Patches:
res_agi.diff uploaded by davevg (license 209)
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r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line
Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro.
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1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
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based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
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revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
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When the XMPP over TLS/SSL connection resets for some reason, it is
wrongly believed as being secured, which makes the re-connection
process endlessly fail. This was reported by mvanbaak in issue #11644.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
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so that paths and filename are writable by asterisk.c without
causing segfaults.
This involves defining the variables as const char *,
and having them point to as static, writable buffer
defined in asterisk.c
On passing, fix some errors in using these variables
in some files in utils/ , and in res/snmp/agent.c
which was redefining a variable without using paths.h
(not applicable to 1.4)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec 2007) | 6 lines
Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor
needs to use this if the user does not have soxmix.
(closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me)
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is below. I should note that the PACKAGE_* macros that asterisk
defines in autoconfig.h are not used anywhere in the tree so
they should just be removed.
/*
* There is some collision collision between netsmp and asterisk names,
* causing build under AST_DEVMODE to fail.
*
* The following PACKAGE_* macros are one place.
* Also netsnmp has an improper check for HAVE_DMALLOC_H, using
* #if HAVE_DMALLOC_H instead of #ifdef HAVE_DMALLOC_H
* As a countermeasure we define it to 0, however this will fail
* when the proper check is implemented.
*/
No
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modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
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r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines
In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.
While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.
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generate loadable and embedded module lists.
Individual Makefiles now are a lot simpler, possibly as simple as this:
-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
MODULE_PREFIX=cdr_
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.
The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).
With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.
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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
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1. When moh is started, we search first in memory to find the class. If we do not
find it in memory, we search realtime instead.
2. When moh is restarted (as in, it had been started on this particular channel, stopped,
and now we're starting it again), if using the "files" mode, then realtime will always
be rechecked. If you are using other modes, however, we will simply reattach to the external
running process which was playing moh earlier in the call. This is a necessary compromise so that
we don't end up with too many background processes.
3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory list. This has the advantage
of not requiring database lookups each time moh is started, but it has the disadvantage of not
truly being realtime.
I have tested this for functionality, and it passes. I also tested this under valgrind and there
are no memory problems reported under typical use.
Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker!
(closes issue #11196, reported and patched by sergee)
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
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After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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through ast_mutex primitives.
To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.
There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).
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This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
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parked extension is provided as an argument.
(closes issue #10803)
Reported by: outtolunc
Patches:
res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237)
- modified by me to work a bit differently ...
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(closes issue #11237)
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r89248 | qwell | 2007-11-13 13:47:45 -0600 (Tue, 13 Nov 2007) | 7 lines
Revert change from revision 67064.
It is documented behavior that if a parking extension already exists while using PARKINGEXTEN,
dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you
must keep this in mind, and handle the failure yourself.
Issue 11237, reported by jon.
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Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line
In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
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the device that made the transfer instead. This makes for much smoother calling back
when queues are involved.
(closes issue #11155, reported by IPetrov)
Tremendous thanks to Russell for pulling me out of my block I was having on this one
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r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines
If someone were to delete the files used by an existing MOH class, and then
issue a reload, further use of that class could result in a crash due to
dividing by zero. This set of changes fixes up some places to prevent this
from happening.
(closes issue #10948)
Reported by: jcomellas
Patches:
res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282)
Additional changes added by me.
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details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
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r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 line
Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check.
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r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 line
closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config,
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Also fixes a few cli messages and some minor formatting.
(closes issue #11001)
Reported by: seanbright
Patches:
newcli.1.patch uploaded by seanbright (license 71)
newcli.2.patch uploaded by seanbright (license 71)
newcli.4.patch uploaded by seanbright (license 71)
newcli.5.patch uploaded by seanbright (license 71)
newcli.6.patch uploaded by seanbright (license 71)
newcli.7.patch uploaded by seanbright (license 71)
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It allows you to configure a prefix for auto-monitor recordings.
(closes issue #6353)
Reported by: ivanfm
Patches:
asterisk_automon_v4.patch uploaded by ivanfm (original patch)
- updated patch:
6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
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Closes issue #10913, reported by tootai, who graciously granted us access
to his Asterisk server, thanks! Daniel, feel free to reopen the bug in
case you can reproduce this on 1.4.
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r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) | 5 lines
Presence packets from a client who's connected with our Jabber ID are
valid, therefore, those clients must be considered as buddies. The resource
string helps us make the distinction between clients.
Closes issue #10707, reported by yusufmotiwala.
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r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) | 5 lines
Prevent Asterisk from crashing when receiving a presence packet
without resource from a buddy that is known to have a resource list.
Revert a change I previously made, where Asterisk could point to a
freed memory location.
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r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1 line
closes issue #10834 ; where a null input to a switch statement results in a hangup; since switch is implemented with extensions, and the default case is implemented with a '.', and the '.' matches 1 or more remaining characters, the case where 0 characters exist isn't matched, and the extension isn't matched, and the goto fails, and a hangup occurs. Now, when a default case is generated, it also generates a single fixed extension that will match a null input. That extension just does a goto to the default extension for that switch. I played with an alternate solution, where I just tack an extra char onto all the patterns and the goto, but not the default case's pattern. Then even a null input will still have at least one char in it. But it made me nervous, having that extra char in , even if that's a pretty secret and low-level issue.
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r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) | 5 lines
Add another sanity check in the AGI read loop. We really don't care about
EAGAIN unless we didn't read an entire line. If there is a newline at the
end if the read buffer, break, because we got the whole thing.
(reported and patched by bmd)
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a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
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r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct 2007) | 6 lines
Fix randomness. save_pos was being set to 0 initially instead of -1, causing it to jump to position 0 when moh started.
(closes issue #10859)
Reported by: jamesgolovich
Patches:
asterisk-mohpos2.diff.txt uploaded by jamesgolovich (license 176)
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r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1 line
This issue sort of closes 10786; All config files support #include with globbing (you know, *,[chars],?,{list,list},etc), so I've updated the AEL system to support this also.
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r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1 line
This closes issue #10788 -- The exact same fixes are made here for the first arg in the for(arg1; arg2; arg3) {} statement, as were done for the 3rd arg. It can now be an assignment that will embedded in a Set() app, or a macro call, or an app call.
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r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1 line
This closes issue #10788 -- the 3rd arg in the for statement is now wrapped in Set() only if there's an '=' in that string. Otherwise, if it begins with '&', then a Macro call is generated; otherwise it is made into an app call. A bit more accomodating, keeps the new guys happy, and the guys with ael-1 code should be happy, too
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | 11 lines
Add a new patch to handle interrupting the fgets() call when using FastAGI.
This version of the patch maintains the original behavior of the code when
not using FastAGI.
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24)
Slight mods by me
Tested by: juggie, festr
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r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) | 5 lines
Handle the case where there are multiple dynamic features with the same digit
mapping, but won't always match the activated on/by access controls. In that
case, the code needs to keep trying features for a match.
(reported by Atis on the asterisk-dev list, patched by me)
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Tilghman Lesher
David Vossel
Jeff Peeler
Russell Bryant
Terry Wilson
Kevin P. Fleming
Richard Mudgett
Olle Johansson
Philippe Sultan
Sean Bright
Michiel van Baak
Matthias Nick
Tzafrir Cohen
Gavin Henry
Mark Michelson
David Brooks
Joshua Colp
Eliel C. Sardanons
Steve Murphy
Matthew Nicholson
Doug Bailey
Brett Bryant
Jason Parker
Donny Kavanagh
Luigi Rizzo