[svn-commits] russell: branch group/bindings r204011 - in /team/group/bindings: ./ apps/ bu...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jun 29 10:03:16 CDT 2009
Author: russell
Date: Mon Jun 29 10:03:10 2009
New Revision: 204011
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=204011
Log:
resolve, reset
Added:
team/group/bindings/cdr/cdr_syslog.c
- copied unchanged from r203985, trunk/cdr/cdr_syslog.c
team/group/bindings/configs/cdr_syslog.conf.sample
- copied unchanged from r203985, trunk/configs/cdr_syslog.conf.sample
Modified:
team/group/bindings/ (props changed)
team/group/bindings/CHANGES
team/group/bindings/UPGRADE.txt
team/group/bindings/apps/app_chanspy.c
team/group/bindings/build_tools/menuselect-deps.in
team/group/bindings/channels/chan_dahdi.c
team/group/bindings/channels/sig_pri.c
team/group/bindings/configure
team/group/bindings/configure.ac
team/group/bindings/main/app.c
team/group/bindings/main/file.c
Propchange: team/group/bindings/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/bindings/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/bindings/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jun 29 10:03:10 2009
@@ -1,1 +1,1 @@
-/trunk:1-203794
+/trunk:1-204010
Modified: team/group/bindings/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/CHANGES?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/CHANGES (original)
+++ team/group/bindings/CHANGES Mon Jun 29 10:03:10 2009
@@ -50,16 +50,19 @@
Applications
------------
* Added progress option to the app_dial D() option. When progress DTMF is
- present, those values are sent immediatly upon receiving a PROGRESS message
+ present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
* Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
* Added c() option to app_chanspy. This option allows custom DTMF to be set
- to cycle through the next avaliable channel. By default this is still '*'.
+ to cycle through the next available channel. By default this is still '*'.
* Added x() option to app_chanspy. This option allows DTMF to be set to
exit the application.
* The Voicemail application has been improved to automatically ignore messages
that only contain silence.
+ * The ChanSpy application now has the 's' option, which makes the application
+ automatically exit once it hits a point where no more channels are available
+ to spy on.
Dialplan Functions
------------------
@@ -84,7 +87,7 @@
The possible values are:
- on - normal mode (the echo canceller is actually reinitalized)
+ on - normal mode (the echo canceller is actually reinitialized)
off - disabled
fax - FAX/data mode (NLP disabled if possible, otherwise completely
disabled)
@@ -163,12 +166,6 @@
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
-Logger
-------
- * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
- users of this channel in the tree have been converted to LOG_NOTICE or removed
- (in cases where the same message was already generated to another channel).
-
Channel Event Logging
---------------------
* A new interface, CEL, is introduced here. CEL logs single events, much like
@@ -185,6 +182,8 @@
linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
etc are performed. Thus the peices of CDR can be grouped into multilegged sets.
* Multiple files and formats can now be specified in cdr_custom.conf.
+ * cdr_syslog has been added which allows CDRs to be written directly to syslog.
+ See configs/cdr_syslog.conf.sample for more information.
Calendaring for Asterisk
------------------------
@@ -500,12 +499,8 @@
as well as periodically updating the IP address. These properties allow for
better performance as well as recovery in the event of an IP change.
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
- load/reload of large numbers of peers/users by ~40x (for large lists of peers.
- Initially, we saw 4x improvement in call setup/destruction, but at the time
- of merging, this gain has disappeared; further research will be done to try
- and restore this performance improvement. Astobj2 refcounting is now used
- for users, peers, and dialogs. Users are encouraged to assist in regression
- testing and problem reporting!
+ load/reload of large numbers of peers/users by ~40x (for large lists of peers).
+ These changes also provide performance improvements for call setup and tear down.
* Added ability to specify registration expiry time on a per registration basis in
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
@@ -1141,11 +1136,11 @@
--------------
* Added rotatestrategy option to logger.conf, along with two new options:
"timestamp" which will use the time to name the logger files instead of
- sequence number; and "rotate", which rotates the names of the logfiles,
+ sequence number; and "rotate", which rotates the names of the log files,
similar to the way syslog rotates files.
* Added exec_after_rotate option to logger.conf, which allows a system
command to be run after rotation. This is primarily useful with
- rotatestrategry=rotate, to allow a limit on the number of logfiles kept
+ rotatestrategy=rotate, to allow a limit on the number of log files kept
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
Modified: team/group/bindings/UPGRADE.txt
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/UPGRADE.txt?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/UPGRADE.txt (original)
+++ team/group/bindings/UPGRADE.txt Mon Jun 29 10:03:10 2009
@@ -20,11 +20,16 @@
From 1.6.2 to 1.6.3:
+* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
+ users of this channel in the tree have been converted to LOG_NOTICE or removed
+ (in cases where the same message was already generated to another channel).
+
* The usage of RTP inside of Asterisk has now become modularized. This means
the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
If you are not using autoload=yes in modules.conf you will need to ensure
it is set to load. If not, then any module which uses RTP (such as chan_sip)
will not be able to send or receive calls.
+
* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
remains. It now exists within app_chanspy.c and retains the exact same
functionality as before.
Modified: team/group/bindings/apps/app_chanspy.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/apps/app_chanspy.c?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/apps/app_chanspy.c (original)
+++ team/group/bindings/apps/app_chanspy.c Mon Jun 29 10:03:10 2009
@@ -134,6 +134,9 @@
</option>
<option name="o">
<para>Only listen to audio coming from this channel.</para>
+ </option>
+ <option name="s">
+ <para>Stop when no more channels are left to spy on.</para>
</option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
@@ -266,6 +269,9 @@
<option name="o">
<para>Only listen to audio coming from this channel.</para>
</option>
+ <option name="s">
+ <para>Stop when there are no more extensions left to spy on.</para>
+ </option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
numeric extension in the current context or the context
@@ -349,6 +355,7 @@
OPTION_DTMF_EXIT = (1 << 14), /* Set DTMF to exit, added for DAHDIScan integration */
OPTION_DTMF_CYCLE = (1 << 15), /* Custom DTMF for cycling next avaliable channel, (default is '*') */
OPTION_DAHDI_SCAN = (1 << 16), /* Scan groups in DAHDIScan mode */
+ OPTION_STOP = (1 << 17),
};
enum {
@@ -373,6 +380,7 @@
AST_APP_OPTION_ARG('r', OPTION_RECORD, OPT_ARG_RECORD),
AST_APP_OPTION_ARG('e', OPTION_ENFORCED, OPT_ARG_ENFORCED),
AST_APP_OPTION('o', OPTION_READONLY),
+ AST_APP_OPTION('s', OPTION_STOP),
AST_APP_OPTION('X', OPTION_EXIT),
AST_APP_OPTION('s', OPTION_NOTECH),
AST_APP_OPTION_ARG('n', OPTION_NAME, OPT_ARG_NAME),
@@ -956,6 +964,9 @@
if (res == -1 || ast_check_hangup(chan))
break;
+ if (ast_test_flag(flags, OPTION_STOP) && !next_autochan) {
+ break;
+ }
}
exit:
Modified: team/group/bindings/build_tools/menuselect-deps.in
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/build_tools/menuselect-deps.in?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/build_tools/menuselect-deps.in (original)
+++ team/group/bindings/build_tools/menuselect-deps.in Mon Jun 29 10:03:10 2009
@@ -51,6 +51,7 @@
OPENSSL=@PBX_OPENSSL@
SUPPSERV=@PBX_SUPPSERV@
SWIG=@PBX_SWIG@
+SYSLOG=@PBX_SYSLOG@
TONEZONE=@PBX_TONEZONE@
UNIXODBC=@PBX_UNIXODBC@
USB=@PBX_USB@
Modified: team/group/bindings/channels/chan_dahdi.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/channels/chan_dahdi.c?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/channels/chan_dahdi.c (original)
+++ team/group/bindings/channels/chan_dahdi.c Mon Jun 29 10:03:10 2009
@@ -15667,7 +15667,7 @@
}
/*< \todo why check for the pseudo in the per-channel section.
* Any actual use for manual setup of the pseudo channel? */
- if (!found_pseudo && reload == 0) {
+ if (!found_pseudo && reload != 1) {
/* use the default configuration for a channel, so
that any settings from real configured channels
don't "leak" into the pseudo channel config
Modified: team/group/bindings/channels/sig_pri.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/channels/sig_pri.c?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/channels/sig_pri.c (original)
+++ team/group/bindings/channels/sig_pri.c Mon Jun 29 10:03:10 2009
@@ -1811,15 +1811,12 @@
}
p->digital = IS_DIGITAL(ast->transfercapability);
- /* Add support for exclusive override */
- if (p->priexclusive)
+
+ /* Should the picked channel be used exclusively? */
+ if (p->priexclusive || p->pri->nodetype == PRI_NETWORK) {
exclusive = 1;
- else {
- /* otherwise, traditional behavior */
- if (p->pri->nodetype == PRI_NETWORK)
- exclusive = 0;
- else
- exclusive = 1;
+ } else {
+ exclusive = 0;
}
pri_sr_set_channel(sr, PVT_TO_CHANNEL(p), exclusive, 1);
Modified: team/group/bindings/configure.ac
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/configure.ac?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/configure.ac (original)
+++ team/group/bindings/configure.ac Mon Jun 29 10:03:10 2009
@@ -1877,6 +1877,8 @@
AC_SUBST([GENERIC_ODBC_INCLUDE])
AC_SUBST([PBX_GENERIC_ODBC])
+PBX_SYSLOG=0
+
if test "${ac_cv_header_syslog_h}" = "yes"; then
# syslog facilities
AST_C_DEFINE_CHECK([SYSLOG_FACILITY_LOG_AUTH], [LOG_AUTH], [syslog.h])
@@ -1890,7 +1892,10 @@
AST_C_DEFINE_CHECK([SYSLOG_FACILITY_LOG_NEWS], [LOG_NEWS], [syslog.h])
AST_C_DEFINE_CHECK([SYSLOG_FACILITY_LOG_SYSLOG], [LOG_SYSLOG], [syslog.h])
AST_C_DEFINE_CHECK([SYSLOG_FACILITY_LOG_UUCP], [LOG_UUCP], [syslog.h])
-fi
+ PBX_SYSLOG=1
+fi
+
+AC_SUBST([PBX_SYSLOG])
AC_CONFIG_FILES([build_tools/menuselect-deps makeopts channels/h323/Makefile])
AST_CHECK_MANDATORY
Modified: team/group/bindings/main/app.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/main/app.c?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/main/app.c (original)
+++ team/group/bindings/main/app.c Mon Jun 29 10:03:10 2009
@@ -700,6 +700,7 @@
struct ast_dsp *sildet = NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
+ int olddspsilence = 0;
int rfmt = 0;
struct ast_silence_generator *silgen = NULL;
char prependfile[80];
@@ -826,7 +827,10 @@
if (maxsilence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
- totalsilence += dspsilence;
+ if (olddspsilence > dspsilence) {
+ totalsilence += olddspsilence;
+ }
+ olddspsilence = dspsilence;
if (dspsilence > maxsilence) {
/* Ended happily with silence */
@@ -906,6 +910,10 @@
if (!prepend) {
/* Reduce duration by a total silence amount */
+ if (olddspsilence <= dspsilence) {
+ totalsilence += dspsilence;
+ }
+
if (totalsilence > 0)
*duration -= (totalsilence - 200) / 1000;
if (*duration < 0) {
Modified: team/group/bindings/main/file.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/group/bindings/main/file.c?view=diff&rev=204011&r1=204010&r2=204011
==============================================================================
--- team/group/bindings/main/file.c (original)
+++ team/group/bindings/main/file.c Mon Jun 29 10:03:10 2009
@@ -1227,7 +1227,14 @@
} else {
res = fr->subclass;
if (strchr(forward, res)) {
+ int eoftest;
ast_stream_fastforward(c->stream, skip_ms);
+ eoftest = fgetc(c->stream->f);
+ if (feof(c->stream->f)) {
+ ast_stream_rewind(c->stream, skip_ms);
+ } else {
+ ungetc(eoftest, c->stream->f);
+ }
} else if (strchr(reverse, res)) {
ast_stream_rewind(c->stream, skip_ms);
} else if (strchr(breakon, res)) {
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