Tilghman Lesher [Mon, 7 Jun 2010 19:52:39 +0000 (19:52 +0000)]
Seems strange (and the code backs up) that if the max and min of a statistic is expressed as a double, the last value would not also need to be a double.
Richard Mudgett [Mon, 7 Jun 2010 15:51:39 +0000 (15:51 +0000)]
Suppress warning in waitstream_core().
Suppress the warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC
in file.c:waitstream_core().
Tilghman Lesher [Sat, 5 Jun 2010 17:55:28 +0000 (17:55 +0000)]
Fix crash in DTMF detection.
What I did not originally see in my previous commit was that even though the
next digit could be detected before the previous was considered ended, the
detection of the next digit effectively ends the detection of the previous.
Therefore, the length moves in lockstep with the digit, and no separate counter
is needed for the length alone.
Changes.
1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is 500ms.
Asterisk was starting the retransmission timers at T1*2 which shouldn't
cause any problems, but is not RFC compliant.
2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction,
if the retransmit timer fires while in the proceeding state that
the request must be retransmitted. Asterisk currently ack's
requests for both INVITE and non-INVITE transactions when a
1XX response is received, this patch changes this for non-INVITE requests.
3. The 'registerattempts' option in sip.conf is supposed to set
how many registry attempts will be made before giving up. When
this option is set to 1, I would expect only one registry attempt
to be made before stopping because of a failure, but instead two are
made. In my opinion this is not expected behavior. This option does
not indicate that these are re-attempts. The logic behind this option
has been changed to only attempt registers the exact number of times
this option is set to. If this option is 0, it still continues to
re-attempt the registration forever.
Richard Mudgett [Fri, 4 Jun 2010 14:45:03 +0000 (14:45 +0000)]
Incoming overlap dialing no longer works after sig_pri extraction.
The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
Leif Madsen [Thu, 3 Jun 2010 18:53:24 +0000 (18:53 +0000)]
Update UPGRADE.txt and CHANGE for CDR functionality changes.
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
Russell Bryant [Wed, 2 Jun 2010 22:46:37 +0000 (22:46 +0000)]
try to fix some random chan_h323 compilation failures
After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
Russell Bryant [Wed, 2 Jun 2010 21:41:54 +0000 (21:41 +0000)]
Ensure the -Wno-strict-aliasing flag makes it, even if ASTCFLAGS has been specified.
When ASTCFLAGS was specified with the make command, Makefile.rules was using
the specified value from the command line and not the one here, making it so this
flag would go missing.
Richard Mudgett [Wed, 2 Jun 2010 21:05:32 +0000 (21:05 +0000)]
Add ETSI Call Waiting support.
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Mark Michelson [Wed, 2 Jun 2010 18:13:18 +0000 (18:13 +0000)]
Prevent use of uninitialized values.
Two struct sockaddr_ins are created when applying directmedia
host access rules. The addresses of these are passed to the RTP
engine to be filled in. However, the RTP engine inspects the fields
of the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
Richard Mudgett [Wed, 2 Jun 2010 18:10:15 +0000 (18:10 +0000)]
Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Jeff Peeler [Wed, 2 Jun 2010 17:29:35 +0000 (17:29 +0000)]
Fix infinite loop when loading codec speex
This changes the sample slinear frame data to contain non-zero data so that
translation calculations for speex works when preprocessing and VAD is turned
on. The encoder expects samples to be returned, but when attempted with the
mentioned two options and silent sample frames everything was discarded.
Richard Mudgett [Wed, 2 Jun 2010 16:14:12 +0000 (16:14 +0000)]
Add ETSI Explicit Call Transfer (ECT) support.
Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Tilghman Lesher [Tue, 1 Jun 2010 16:41:00 +0000 (16:41 +0000)]
Eliminate stale manager events after a set interval, even if AMI clients don't query for them.
Actions (or failures to act) by external clients should not cause memory leaks
in Asterisk, especially when those continued leaks could cause Asterisk to
misbehave later.
Terry Wilson [Fri, 28 May 2010 22:54:03 +0000 (22:54 +0000)]
Fix ical library handling (again)
Newer versions of libical (which we require) store the header file in a
libical/ subfolder and include an ical.h file that does a #warning for
deprecation and then #includes <libical/ical.h>. Since we now test for
libical/ical.h, we can change the #includes back to <libical/ical.h> and
remove the test which specifically adds /usr/include/libical as an include
directory.
Use sigaction for signals which should persist past the initial trigger, not signal.
If you call signal() in a Solaris signal handler, instead of just resetting
the signal handler, it causes the signal to refire, because the signal is not
marked as handled prior to the signal handler being called. This effectively
causes Solaris to immediately exceed the threadstack in recursive signal
handlers and crash.
David Vossel [Wed, 26 May 2010 19:46:49 +0000 (19:46 +0000)]
do all sip registry parsing before transmit_register
This patch breaks up every part of the sip registry string during
config parsing and removes all parsing from transmit_register().
Thanks to Nick_Lewis for contributing this patch!
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657)
chan_sip.c.patch uploaded by Nick Lewis (license 657)
chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657)
nicklewispatch.diff uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
Terry Wilson [Wed, 26 May 2010 05:33:11 +0000 (05:33 +0000)]
Ensure that libneon > 0.29.0 is installed for res_calendar_ews
This uses a modified version of pabelanger's patch that checks for NTLM support
instead, which was added in 0.29.0 which is what is required for
res_calendar_ews.
Mark Michelson [Tue, 25 May 2010 20:59:04 +0000 (20:59 +0000)]
Properly use peer's outboundproxy for outbound REGISTERs.
The logic used in transmit_register to get the outboundproxy for a peer
was flawed since this value would be overridden shortly afterwards when
create_addr was called.
In addition, this also fixes some logic used when parsing users.conf so
that the peer name is placed in the internally-generated register string
so that an outboundproxy set in the Asterisk GUI will be used for outbound
REGISTERs.
David Vossel [Mon, 24 May 2010 19:42:54 +0000 (19:42 +0000)]
reverses incorrect logic introduced by r243200
The decoding of the replace_id did not need to be broken
up in this instance. This was brought to my attention
again because it caused a segfault when the from or to
tags were not present in the "Replaces" header.
Terry Wilson [Mon, 24 May 2010 19:06:40 +0000 (19:06 +0000)]
Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:
Terry Wilson [Mon, 24 May 2010 18:21:20 +0000 (18:21 +0000)]
Calendaring support for Exchange Server 2007+ via EWS
This commit adds support for calendaring with Exchange Server 2007+ via
Exchange Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature.
Richard Mudgett [Fri, 21 May 2010 22:46:52 +0000 (22:46 +0000)]
Channel initialization failure causes crashes.
__ast_channel_alloc_ap() has several points in the initialization of a new
channel structure where it could fail. Since the channel structure is now
an ao2 object, the destructor callback needs to be able to handle clean up
when the structure setup is incomplete.
Problems corrected:
1) Failing to setup the alertpipe would not unreference the structure but
free it directly. Doing this to an ao2_object is very bad.
2) File descriptors need to be initialized to -1 before a construction
failure could occur so the destructor will not close unopened descriptors.
3) The destructor needs to check that the string field has been
initialized before using any string field values. Crashes expected.
4) The destructor should not notify devstate if the device name is empty.
It is a waste of cycles and a couple ERROR log messages are generated.
projects / thirdparty / asterisk.git / log
