[svn-commits] newtonr: branch 11 r405792 - in /branches/11: ./ apps/ channels/ configs/ doc...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jan 17 09:40:41 CST 2014
Author: newtonr
Date: Fri Jan 17 09:40:37 2014
New Revision: 405792
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=405792
Log:
Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
........
Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Modified:
branches/11/ (props changed)
branches/11/apps/app_queue.c
branches/11/apps/app_transfer.c
branches/11/channels/chan_iax2.c
branches/11/channels/chan_sip.c
branches/11/configs/sip.conf.sample
branches/11/doc/asterisk.8
branches/11/main/features.c
Propchange: branches/11/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: branches/11/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/apps/app_queue.c?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/apps/app_queue.c (original)
+++ branches/11/apps/app_queue.c Fri Jan 17 09:40:37 2014
@@ -230,7 +230,7 @@
<para>Will run a macro on the calling party's channel once they are connected to a queue member.</para>
</parameter>
<parameter name="gosub">
- <para>Will run a gosub on the calling party's channel once they are connected to a queue member.</para>
+ <para>Will run a gosub on the called party's channel (the queue member) once the parties are connected.</para>
</parameter>
<parameter name="rule">
<para>Will cause the queue's defaultrule to be overridden by the rule specified.</para>
Modified: branches/11/apps/app_transfer.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/apps/app_transfer.c?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/apps/app_transfer.c (original)
+++ branches/11/apps/app_transfer.c Fri Jan 17 09:40:37 2014
@@ -54,7 +54,7 @@
<description>
<para>Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
- an incoming call with the same channel technology will be transfered.
+ an incoming call with the same channel technology will be transferred.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.</para>
<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable>
Modified: branches/11/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/channels/chan_iax2.c?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/channels/chan_iax2.c (original)
+++ branches/11/channels/chan_iax2.c Fri Jan 17 09:40:37 2014
@@ -5602,7 +5602,7 @@
unlock_both(callno0, callno1);
return AST_BRIDGE_FAILED_NOWARN;
}
- /* check if transfered and if we really want native bridging */
+ /* check if transferred and if we really want native bridging */
if (!transferstarted && !ast_test_flag64(iaxs[callno0], IAX_NOTRANSFER) && !ast_test_flag64(iaxs[callno1], IAX_NOTRANSFER)) {
/* Try the transfer */
if (iax2_start_transfer(callno0, callno1, (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1)) ||
Modified: branches/11/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/channels/chan_sip.c?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/channels/chan_sip.c (original)
+++ branches/11/channels/chan_sip.c Fri Jan 17 09:40:37 2014
@@ -6292,7 +6292,7 @@
} else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
ast_string_field_set(p, fromdomain, ast_var_value(current));
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
- /* This is a transfered call */
+ /* This is a transferred call */
p->options->transfer = 1;
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
/* This is the referrer */
@@ -6349,7 +6349,7 @@
if (referer) {
if (sipdebug)
- ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
+ ast_debug(3, "Call for %s transferred by %s\n", p->username, referer);
snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
} else
snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
@@ -24566,7 +24566,7 @@
/* EventID for each transfer... EventID is basically the REFER cseq
- We are getting notifications on a call that we transfered
+ We are getting notifications on a call that we transferred
We should hangup when we are getting a 200 OK in a sipfrag
Check if we have an owner of this event */
Modified: branches/11/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/configs/sip.conf.sample (original)
+++ branches/11/configs/sip.conf.sample Fri Jan 17 09:40:37 2014
@@ -395,6 +395,9 @@
; support it. This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]
+
+;textsupport=no ; Support for ITU-T T.140 realtime text.
+ ; The default value is "no".
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
Modified: branches/11/doc/asterisk.8
URL: http://svnview.digium.com/svn/asterisk/branches/11/doc/asterisk.8?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/doc/asterisk.8 (original)
+++ branches/11/doc/asterisk.8 Fri Jan 17 09:40:37 2014
@@ -16,27 +16,27 @@
\fBasterisk\fR \kx
.if (\nx>(\n(.l/2)) .nr x (\n(.l/5)
'in \n(.iu+\nxu
-[\fB-BcdfFghiImnpqRtTvVW\fR] [\fB-C \fR\fIfile\fR] [\fB-e \fR\fImemory\fR] [\fB-G \fR\fIgroup\fR] [\fB-L \fR\fIloadaverage\fR] [\fB-M \fR\fIvalue\fR] [\fB-U \fR\fIuser\fR] [\fB-s \fR\fIsocket-file\fR]
+[\fB\-BcdfFghiImnpqRtTvVW\fR] [\fB\-C \fR\fIfile\fR] [\fB\-e \fR\fImemory\fR] [\fB\-G \fR\fIgroup\fR] [\fB\-L \fR\fIloadaverage\fR] [\fB\-M \fR\fIvalue\fR] [\fB\-U \fR\fIuser\fR] [\fB\-s \fR\fIsocket\-file\fR]
+'in \n(.iu\-\nxu
+.ad b
+'hy
+'nh
+.fi
+.ad l
+\fBasterisk \-r\fR \kx
+.if (\nx>(\n(.l/2)) .nr x (\n(.l/5)
+'in \n(.iu+\nxu
+[\fB\-v\fR] [\fB\-d\fR] [\fB\-x \fR\fIcommand\fR]
'in \n(.iu-\nxu
.ad b
'hy
'nh
.fi
.ad l
-\fBasterisk -r\fR \kx
+\fBasterisk \-R\fR \kx
.if (\nx>(\n(.l/2)) .nr x (\n(.l/5)
'in \n(.iu+\nxu
-[\fB-v\fR] [\fB-d\fR] [\fB-x \fR\fIcommand\fR]
-'in \n(.iu-\nxu
-.ad b
-'hy
-'nh
-.fi
-.ad l
-\fBasterisk -R\fR \kx
-.if (\nx>(\n(.l/2)) .nr x (\n(.l/5)
-'in \n(.iu+\nxu
-[\fB-v\fR] [\fB-d\fR] [\fB-x \fR\fIcommand\fR]
+[\fB\-v\fR] [\fB\-d\fR] [\fB\-x \fR\fIcommand\fR]
'in \n(.iu-\nxu
.ad b
'hy
@@ -52,12 +52,12 @@
.PP
At start, Asterisk reads the /etc/asterisk/asterisk.conf main configuration
file and locates the rest of the configuration files from the configuration
-in that file. The -C option specifies an alternate main configuration file.
+in that file. The \-C option specifies an alternate main configuration file.
Virtually all aspects of the operation of asterisk's configuration files
can be found in the sample configuration files. The format for those files
is generally beyond the scope of this man page.
.PP
-When running with \fB-c\fR, \fB-r\fR or \fB-R\fR
+When running with \fB\-c\fR, \fB\-r\fR or \fB\-R\fR
options, Asterisk supplies a powerful command line, including command
completion, which may be used to monitors its status, perform a variety
of administrative actions and even explore the applications that are
@@ -70,26 +70,26 @@
\*(T<\fB\-R\fR\*(T> connects to an existing Asterisk instance through
a remote console.
.TP
--B
+\-B
Force the background of the terminal to be black, in order for
terminal colors to show up properly. Equivalent to
\*(T<\fBforceblackbackground = yes\fR\*(T> in
\*(T<\fIasterisk.conf\fR\*(T>. See also
\*(T<\fB\-n\fR\*(T> and \*(T<\fB\-W\fR\*(T>.
.TP
--C \fIfile\fR
+\-C \fIfile\fR
Use \*(T<\fIfile\fR\*(T> as master configuration file
instead of the default, /etc/asterisk/asterisk.conf
.TP
--c
+\-c
Provide a control console on the calling terminal. The
console is similar to the remote console provided by
\*(T<\fB\-r\fR\*(T>. Specifying this option implies
-\fB-f\fR and will cause asterisk to no longer
+\fB\-f\fR and will cause asterisk to no longer
fork or detach from the controlling terminal. Equivalent
to \*(T<\fBconsole = yes\fR\*(T> in \*(T<\fIasterisk.conf\fR\*(T>.
.TP
--d
+\-d
Enable extra debugging statements. This parameter may be used several
times, and each increases the debug level. Equivalent to \*(T<\fBdebug = \fR\*(T>\fInum\fR
in \*(T<\fIasterisk.conf\fR\*(T> to explicitly set the initian debug
@@ -104,62 +104,62 @@
Equivalent to \*(T<\fBminmemfree = \fR\*(T>\fImemory\fR in
\*(T<\fIasterisk.conf\fR\*(T>.
.TP
--f
+\-f
Do not fork or detach from controlling terminal. Overrides any
-preceding specification of \fB-F\fR on the command line.
+preceding specification of \fB\-F\fR on the command line.
Equivalent to \*(T<\fBnofork = yes\fR\*(T> in \*(T<\fIasterisk.conf\fR\*(T>.
See also \*(T<\fB\-c\fR\*(T>.
.TP
--F
+\-F
Always fork and detach from controlling terminal. Overrides any
-preceding specification of \fB-f\fR on the command line.
+preceding specification of \fB\-f\fR on the command line.
May also be used to prevent \*(T<\fB\-d\fR\*(T> and \*(T<\fB\-v\fR\*(T> to imply
no forking. Equivalent to \*(T<\fBalwaysfork = yes\fR\*(T> in \*(T<\fIasterisk.conf\fR\*(T>.
.TP
--g
+\-g
Remove resource limit on core size, thus forcing Asterisk to dump
core in the unlikely event of a segmentation fault or abort signal.
\fBNOTE:\fR in some cases this may be incompatible
-with the \fB-U\fR or \fB-G\fR flags.
-.TP
--G \fIgroup\fR
+with the \fB\-U\fR or \fB\-G\fR flags.
+.TP
+\-G \fIgroup\fR
Run as group \fIgroup\fR instead of the
calling group. \fBNOTE:\fR this requires substantial work
to be sure that Asterisk's environment has permission to write
the files required for its operation, including logs, its comm
socket, the asterisk database, etc.
.TP
--h
+\-h
Provide brief summary of command line arguments and terminate.
.TP
--i
+\-i
Prompt user to intialize any encrypted private keys for IAX2
secure authentication during startup.
.TP
--I
+\-I
Enable internal timing if DAHDI timing is available.
The default behaviour is that outbound packets are phase locked
to inbound packets. Enabling this switch causes them to be
locked to the internal DAHDI timer instead.
.TP
--L \fIloadaverage\fR
+\-L \fIloadaverage\fR
Limits the maximum load average before rejecting new calls. This can
be useful to prevent a system from being brought down by terminating
too many simultaneous calls.
.TP
--m
+\-m
Temporarily mutes output to the console and logs. To return to normal,
use \fBlogger mute\fR.
.TP
--M \fIvalue\fR
+\-M \fIvalue\fR
Limits the maximum number of calls to the specified value. This can
be useful to prevent a system from being brought down by terminating
too many simultaneous calls.
.TP
--n
+\-n
Disable ANSI colors even on terminals capable of displaying them.
.TP
--p
+\-p
If supported by the operating system (and executing as root),
attempt to run with realtime priority for increased performance and
responsiveness within the Asterisk process, at the expense of other
@@ -170,77 +170,77 @@
running or is killed, \fBasterisk\fR will slow down to
normal process priority, to avoid locking up the machine.
.TP
--q
+\-q
Reduce default console output when running in conjunction with
-console mode (\fB-c\fR).
-.TP
--r
+console mode (\fB\-c\fR).
+.TP
+\-r
Instead of running a new Asterisk process, attempt to connect
to a running Asterisk process and provide a console interface
for controlling it.
.TP
--R
-Much like \fB-r\fR. Instead of running a new Asterisk process, attempt to connect
+\-R
+Much like \fB\-r\fR. Instead of running a new Asterisk process, attempt to connect
to a running Asterisk process and provide a console interface
for controlling it. Additionally, if connection to the Asterisk
process is lost, attempt to reconnect for as long as 30 seconds.
.TP
--s \fIsocket file name\fR
-In combination with \fB-r\fR, connect directly to a specified
+\-s \fIsocket file name\fR
+In combination with \fB\-r\fR, connect directly to a specified
Asterisk server socket.
.TP
--t
+\-t
When recording files, write them first into a temporary holding directory,
then move them into the final location when done.
.TP
--T
+\-T
Add timestamp to all non-command related output going to the console
when running with verbose and/or logging to the console.
.TP
--U \fIuser\fR
+\-U \fIuser\fR
Run as user \fIuser\fR instead of the
calling user. \fBNOTE:\fR this requires substantial work
to be sure that Asterisk's environment has permission to write
the files required for its operation, including logs, its comm
socket, the asterisk database, etc.
.TP
--v
+\-v
Increase the level of verboseness on the console. The more times
-\fB-v\fR is specified, the more verbose the output is.
-Specifying this option implies \fB-f\fR and will cause
+\fB\-v\fR is specified, the more verbose the output is.
+Specifying this option implies \fB\-f\fR and will cause
asterisk to no longer fork or detach from the controlling terminal.
-This option may also be used in conjunction with \fB-r\fR
-and \fB-R\fR.
+This option may also be used in conjunction with \fB\-r\fR
+and \fB\-R\fR.
Note: This always sets the verbose level in the asterisk process,
even if it is running in the background. This will affect the size
of your log files.
.TP
--V
+\-V
Display version information and exit immediately.
.TP
--W
+\-W
Display colored terminal text as if the background were white
or otherwise light in color. Normally, terminal text is displayed
as if the background were black or otherwise dark in color.
.TP
--x \fIcommand\fR
+\-x \fIcommand\fR
Connect to a running Asterisk process and execute a command on
a command line, passing any output through to standard out and
then terminating when the command execution completes. Implies
-\fB-r\fR when \fB-R\fR is not explicitly
+\fB\-r\fR when \fB\-R\fR is not explicitly
supplied.
.TP
--X
+\-X
Enables executing of includes via \fB#exec\fR directive.
This can be useful if You want to do \fB#exec\fR inside
\*(T<\fIasterisk.conf\fR\*(T>
.SH EXAMPLES
\fBasterisk\fR - Begin Asterisk as a daemon
.PP
-\fBasterisk -vvvgc\fR - Run on controlling terminal
-.PP
-\fBasterisk -rx "core show channels"\fR - Display channels on running server
+\fBasterisk \-vvvgc\fR - Run on controlling terminal
+.PP
+\fBasterisk \-rx "core show channels"\fR - Display channels on running server
.SH BUGS
Bug reports and feature requests may be filed at https://issues.asterisk.org
.SH "SEE ALSO"
Modified: branches/11/main/features.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/main/features.c?view=diff&rev=405792&r1=405791&r2=405792
==============================================================================
--- branches/11/main/features.c (original)
+++ branches/11/main/features.c Fri Jan 17 09:40:37 2014
@@ -2497,7 +2497,7 @@
/*!
* \brief Blind transfer user to another extension
- * \param chan channel to be transfered
+ * \param chan channel to be transferred
* \param peer channel initiated blind transfer
* \param config
* \param code
@@ -2661,7 +2661,7 @@
/*!
* \brief Attended transfer
- * \param chan transfered user
+ * \param chan transferred user
* \param peer person transfering call
* \param config
* \param code
@@ -3845,7 +3845,7 @@
* \details
* Request channel, set channel variables, initiate call,
* check if they want to disconnect, go into loop, check if timeout has elapsed,
- * check if person to be transfered hung up, check for answer break loop,
+ * check if person to be transferred hung up, check for answer break loop,
* set cdr return channel.
*
* \retval Channel Connected channel for transfer.
@@ -4463,7 +4463,7 @@
ast_set_flag(peer_cdr, AST_CDR_FLAG_BRIDGED);
}
}
- /* the DIALED flag may be set if a dialed channel is transfered
+ /* the DIALED flag may be set if a dialed channel is transferred
* and then bridged to another channel. In order for the
* bridge CDR to be written, the DIALED flag must not be
* present. */
@@ -8157,7 +8157,7 @@
*
* Split data, check we aren't bridging with ourself, check valid channel,
* answer call if not already, check compatible channels, setup bridge config
- * now bridge call, if transfered party hangs up return to PBX extension.
+ * now bridge call, if transferred party hangs up return to PBX extension.
*/
static int bridge_exec(struct ast_channel *chan, const char *data)
{
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