* Classes are now tracked past removal from the core container, and module
removal is actively prevented until all references are freed.
* A hanging reference stored in the channel has been removed. This could have
caused a mismatch and the music state not properly cleared, if two or more
reloads occurred between MOH being stopped and MOH being restarted.
* In certain circumstances, duplicate classes were possible.
* A race existed at reload time between a process being killed and the thread
responsible for reading from the related pipe respawning that process.
* Several reference counts have also been corrected. At least one could have
caused deleted classes to stick around forever, consuming resources. This
originally manifested as MOH external processes that were not killed at
reload time.
(closes issue #16279, closes issue #16207)
Reported by: parisioa, dcabot
Patches:
20091202__issue16279__2.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232660 f38db490-d61c-443f-a65b-d21fe96a405b
The manpage lies; it says that the function is in POSIX, but that's only for
ffs(3), not ffsll(3).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232164 f38db490-d61c-443f-a65b-d21fe96a405b
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231692 f38db490-d61c-443f-a65b-d21fe96a405b
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230994 f38db490-d61c-443f-a65b-d21fe96a405b
This patch, originally submitted by jozza, enables custom modules to send actions to AMI
and receive messages from AMI via a hook interface. Included is a simple test module to
illustrate the interface.
(closes issue #14635)
Reported by: jozza
Review: https://reviewboard.asterisk.org/r/412/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227448 f38db490-d61c-443f-a65b-d21fe96a405b
This is a side project I've been poking at this week. The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together. There is a ton of stuff to write about, so this will
just continue to evolve over time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226606 f38db490-d61c-443f-a65b-d21fe96a405b
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
Added documentation on how to create a local git repository from
SVN. This documentation was added via doxygen.
(closes issue #15814)
Reported by: tzafrir
Patches:
git-asterisk-howto uploaded by tzafrir (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225483 f38db490-d61c-443f-a65b-d21fe96a405b
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225445 f38db490-d61c-443f-a65b-d21fe96a405b
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225357 f38db490-d61c-443f-a65b-d21fe96a405b
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225244 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
Isolate frames returned from a DSP instance or codec translator.
The reasoning for these changes are the same as what I wrote in the commit
message for rev 222878.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224932 f38db490-d61c-443f-a65b-d21fe96a405b
Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224225 f38db490-d61c-443f-a65b-d21fe96a405b
This isn't the best way to do this, but it is the easiest. There are some
limitations that are going to need to be addressed at some point with reloads
and when I (or someone else) work on that, then the API can be updated to
handle passing the private config data that the calendar tech modules need in
a better way as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223016 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222880 f38db490-d61c-443f-a65b-d21fe96a405b
Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222761 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222176 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220344 f38db490-d61c-443f-a65b-d21fe96a405b
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
(closes issue #15902)
Reported by: lmadsen
Patches:
mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219895 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219139 f38db490-d61c-443f-a65b-d21fe96a405b
Most importantly, note that a NULL description will cause a
crash, as I just experienced that firsthand.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217158 f38db490-d61c-443f-a65b-d21fe96a405b
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217074 f38db490-d61c-443f-a65b-d21fe96a405b
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216094 f38db490-d61c-443f-a65b-d21fe96a405b
This did not function in the way that was intended, causing more compatibility
issues than it solved. It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215800 f38db490-d61c-443f-a65b-d21fe96a405b
One note on defining _POSIX_C_SOURCE: while this feature test macro
works to require certain behaviors on Linux, it works differently on *BSD
platforms to REMOVE certain API calls that are not in the POSIX specification,
such as vasprintf(3). Thus, defining it while depending upon vasprintf (and
other extensions to the POSIX standard) to be defined is a recipe to ensure
that Asterisk is only buildable on Linux.
Hence, this define which was meant to INCREASE portability, effectively
ensures the opposite.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214863 f38db490-d61c-443f-a65b-d21fe96a405b
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214696 f38db490-d61c-443f-a65b-d21fe96a405b
It's not clear that we actually require XOPEN_VERSION to be 600 or greater
at this time, so skip the check for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212672 f38db490-d61c-443f-a65b-d21fe96a405b
Based on a post on the gcc-help mailing list and some subsequent reading,
we can increase our portability to various platforms by directly defining
the POSIX and X/OPEN API feature sets we wish to have available. This patch
does that, and also includes a double-check to ensure that the system
we are compiling on can actually provide the requested feature sets.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212463 f38db490-d61c-443f-a65b-d21fe96a405b
This patch makes some small changes to handle watchdog timeouts in a better way,
and also uses a 'cleaner' method of including the spandsp header files.
(closes issue #14769)
Reported by: andrew
Patches:
app_fax-20090406.diff uploaded by andrew (license 240)
v1-14769.patch uploaded by dimas (license 88)
Tested by: freh, deti, caspy, dimas, sgimeno, Dovid
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210777 f38db490-d61c-443f-a65b-d21fe96a405b
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208548 f38db490-d61c-443f-a65b-d21fe96a405b
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces the security events API. This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication. These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.
Inside of Asterisk, the events go through the ast_event API. This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.
One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level. Using logger.conf, these log entries may be
sent to a file, or to syslog.
One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip. That will be more complicated and will
be done as its own project as the next phase of security events work.
For more information on the security events framework, see the documentation
generated from doc/tex/. "make asterisk.pdf"
Review: https://reviewboard.asterisk.org/r/273/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206021 f38db490-d61c-443f-a65b-d21fe96a405b
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205696 f38db490-d61c-443f-a65b-d21fe96a405b
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205120 f38db490-d61c-443f-a65b-d21fe96a405b
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204919 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204710 f38db490-d61c-443f-a65b-d21fe96a405b
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204413 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
Also change the default casing of the string contants to lowercase. This really
just saves us from have to lowercase them later when displaying them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203605 f38db490-d61c-443f-a65b-d21fe96a405b
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@202410 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
When the list to be appended is empty, and the list to be appended to is *not*,
AST_LIST_APPEND_LIST would actually cause the target list to become broken,
and no longer have a pointer to its last entry. This patch fixes the problem.
(reported by Stanislaw Pitucha on the asterisk-dev mailing list)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201262 f38db490-d61c-443f-a65b-d21fe96a405b
Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201139 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200656 f38db490-d61c-443f-a65b-d21fe96a405b
This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200519 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
(closes issue #15191)
Reported by: alecdavis
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/262/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199743 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
Increase the size of our thread stack on 64 bit processors.
We were setting the stack size for each thread to 240KB regardless of
architecture, which meant that in some scenarios we actually had less available
stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's __WORDSIZE, which
gives us:
32 bit -> 240KB
64 bit -> 496KB
128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of #asterisk-dev, including
Kevin, Tilghman, and Qwell.
(closes issue #14932)
Reported by: jpiszcz
Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
Fix a typo in the stack size calculation just introduced.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199630 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave unexpected results.
(closes issue #15057)
Reported by: p_lindheimer
Patches:
onhold_trunk.diff uploaded by dvossel (license 671)
pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671)
Tested by: p_lindheimer, dvossel
Review: https://reviewboard.asterisk.org/r/254/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199298 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199051 f38db490-d61c-443f-a65b-d21fe96a405b
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198856 f38db490-d61c-443f-a65b-d21fe96a405b
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198727 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198072 f38db490-d61c-443f-a65b-d21fe96a405b
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197738 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197543 f38db490-d61c-443f-a65b-d21fe96a405b
'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196758 f38db490-d61c-443f-a65b-d21fe96a405b
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195882 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf.
Review: https://reviewboard.asterisk.org/r/244/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194610 f38db490-d61c-443f-a65b-d21fe96a405b
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192362 f38db490-d61c-443f-a65b-d21fe96a405b
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192357 f38db490-d61c-443f-a65b-d21fe96a405b
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192318 f38db490-d61c-443f-a65b-d21fe96a405b
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192279 f38db490-d61c-443f-a65b-d21fe96a405b
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192059 f38db490-d61c-443f-a65b-d21fe96a405b
This patch add a new payload type for information elements, a set
of bit flags. The payload is transported as a 32-bit unsigned integer
but when matching is performed between events and subscribers,
the matching is done by using a bitwise AND instead of numeric value
comparison.
Review: http://reviewboard.asterisk.org/r/242/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191919 f38db490-d61c-443f-a65b-d21fe96a405b
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191785 f38db490-d61c-443f-a65b-d21fe96a405b
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191411 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191177 f38db490-d61c-443f-a65b-d21fe96a405b
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191028 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190725 f38db490-d61c-443f-a65b-d21fe96a405b
Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190545 f38db490-d61c-443f-a65b-d21fe96a405b
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190057 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines
Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h
This allows config.c to compile when linked against uclibc that does not support these parameters
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@189629 f38db490-d61c-443f-a65b-d21fe96a405b
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188342 f38db490-d61c-443f-a65b-d21fe96a405b
The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187491 f38db490-d61c-443f-a65b-d21fe96a405b
This was accomplished using a set of options and the setoption channel callback.
The core calls into the channel driver using these options and the channel driver
either returns success or failure.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187360 f38db490-d61c-443f-a65b-d21fe96a405b
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187269 f38db490-d61c-443f-a65b-d21fe96a405b
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187211 f38db490-d61c-443f-a65b-d21fe96a405b
doxyref.h was created to hold miscellaneous documentation that was not specific
to a part of the code. This file has grown quite a bit so I decided to start
splitting parts of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186953 f38db490-d61c-443f-a65b-d21fe96a405b
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.
(closes issue #12713)
Reported by: davidw
Tested by: file
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186382 f38db490-d61c-443f-a65b-d21fe96a405b
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
Included is a small bugfix to an ast_str helper, but most of these changes
are simply doxygen fixes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185912 f38db490-d61c-443f-a65b-d21fe96a405b
This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here:
Changes:
- Cleanup of some code, fix incorrect doxygen comments
- When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use
- When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space
- When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated
- Don't automatically double the size of each successive pool allocated; it's wasteful
http://reviewboard.digium.com/r/165/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185581 f38db490-d61c-443f-a65b-d21fe96a405b
The doxygen documentation has now been updated to state explicitly that I want
punctuation atthe end of the first sentence in a commit message. :).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184798 f38db490-d61c-443f-a65b-d21fe96a405b
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184762 f38db490-d61c-443f-a65b-d21fe96a405b
When dvossel and I were doing some load testing last week, we noticed that we
could make Asterisk trunk lock up instantly when we started generating a bunch
of calls. The backtraces of locked threads were bizarre, and many were stuck
on an _unlock_ of an rwlock.
The changes are:
1) Fix a number of places where a backtrace would be loaded into an invalid
index of the backtrace array. It's an off by one error, which ends up
writing over the rwlock itself.
2) Ensure that in the array of held locks, we NULL out an index once it is
not being used so that it's not confusing when analyzing its contents.
3) Remove a bunch of logging referring to an rwlock operating being done
with "deep reentrancy". It is normal for _many_ threads to hold a
read lock on an rwlock.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184531 f38db490-d61c-443f-a65b-d21fe96a405b
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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184339 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183436 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@183172 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182847 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182826 f38db490-d61c-443f-a65b-d21fe96a405b
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182525 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181135 f38db490-d61c-443f-a65b-d21fe96a405b
Copied from my review board description:
This is a continuation of the API changes documentation started for describing
changes between releases. Most of the API changes were pretty simple needing
only to be brought to attention via the new "Asterisk API Changes" list.
However, if you see anything that needs further explanation feel free to
supplement what is there. The current method of documenting is to add (in the
header file): \version <ver number> <description of changes> and then to add
the function to the change list in doxyref.h on the AstAPIChanges page. I also
made sure all the functions that were newly added were tagged with \since
1.6.1. I think this is a good habit to start both for the historical aspect as
well as for the future ability to easily add a "New Asterisk API" page.
Review: http://reviewboard.digium.com/r/190/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180719 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180373 f38db490-d61c-443f-a65b-d21fe96a405b
This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180369 f38db490-d61c-443f-a65b-d21fe96a405b
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata().
(closes issue #14279)
Reported by: Marquis
Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671)
Tested by: Marquis, dvossel
Review: http://reviewboard.digium.com/r/177/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180032 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines
This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177732 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
Modify bridging to properly evaluate DTMF after first warning is played
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176708 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@176666 f38db490-d61c-443f-a65b-d21fe96a405b
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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines
fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
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r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines
correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176255 f38db490-d61c-443f-a65b-d21fe96a405b
1) It had numerous coding guidelines violations with regards to formatting.
2) It allocated memory using ast_calloc() that was never freed.
3) It didn't check for failure from the allocation.
4) It used sprintf() and strcat() to build the result, doing zero checking to
prevent writing past the end of the provided buffer.
The function also lacks API documentation, but that has not been addressed in
this commit.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175829 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175508 f38db490-d61c-443f-a65b-d21fe96a405b
Core show locks information involving an ao2_trylock did not
show the function that called ao2_trylock, but would instead
show ao2_trylock as the source of the lock. This is not useful
when trying to debug locking issues.
One bizarre note is that this logic is already in 1.4 but somehow
did not get merged to trunk or the 1.6.X branches.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175121 f38db490-d61c-443f-a65b-d21fe96a405b
The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174945 f38db490-d61c-443f-a65b-d21fe96a405b
This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread. chan_iax2 has been
updated to use this new code. Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.
Related to rev 171452 which resolved the same issues in 1.4.
Code from team/russell/sched_thread2
Review: http://reviewboard.digium.com/r/129/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173858 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines
Parking attempts made to one end of a bridge no longer will hang up due to a
parking failure.
Parking attempts made using either one-touch, or doing either a blind or
assisted transfer to the parking extension now keep up the bridge instead of
hanging up the attempted parked party. Normal causes for the parking attempt
to fail includes the specific specified extension (via PARKINGEXTEN) not being
available or if all the parking spaces are currently in use. To avoid having
to reverse a masquerade park_space_reserve was made to provide foresight if
a parking attempt will succeed and if so reserve the parking space.
(closes issue #13494)
Reported by: mdu113
Reviewed by Russell: http://reviewboard.digium.com/r/133/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173500 f38db490-d61c-443f-a65b-d21fe96a405b
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
Reported by: oej
Tested by: Corydon76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173130 f38db490-d61c-443f-a65b-d21fe96a405b
reviewboard this before committing again...
reopened 11583 until all Russell's issues are
resolved.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172929 f38db490-d61c-443f-a65b-d21fe96a405b
to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172890 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172580 f38db490-d61c-443f-a65b-d21fe96a405b
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172318 f38db490-d61c-443f-a65b-d21fe96a405b
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172268 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172063 f38db490-d61c-443f-a65b-d21fe96a405b
When the second part of this macro is written as 0[a] instead of a[0], it will
force a failure if the macro is used on a C++ object that overloads the []
operator.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@170943 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines
AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is
whether autoconf detected a static initializer value. This fixes rwlocks
on all such platforms (mainly, Mac OS X).
(closes issue #13767)
Reported by: jcovert
Patches:
20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
Tested by: jcovert, Corydon76
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169944 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@169438 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
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r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166665 f38db490-d61c-443f-a65b-d21fe96a405b
This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes. It is currently used in AMI to
address the issue that has been reported. However, there are probably a few
other places where this could be used.
(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166282 f38db490-d61c-443f-a65b-d21fe96a405b
