Documentation: doc fixes across various parts of the code for ASTERISK issues 23061...

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
........

Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_queue.c b/apps/app_queue.c
index e1c8a00..97ea60c 100644 (file)
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -244,7 +244,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                                <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>

diff --git a/apps/app_transfer.c b/apps/app_transfer.c
index 3ed66f1..1b81082 100644 (file)
--- a/apps/app_transfer.c
+++ b/apps/app_transfer.c
@@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                <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>

diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index dcbd210..bb8c9eb 100644 (file)
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -5468,7 +5468,7 @@ static enum ast_bridge_result iax2_bridge(struct ast_channel *c0, struct ast_cha
                        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)) ||

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index cc17ad3..0d0f446 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6417,7 +6417,7 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
                } else if (!strcmp(ast_var_name(current), "SIPFROMDOMAIN")) {
                        ast_string_field_set(p, fromdomain, ast_var_value(current));
                } else if (!strcmp(ast_var_name(current), "SIPTRANSFER")) {
-                       /* This is a transfered call */
+                       /* This is a transferred call */
                        p->options->transfer = 1;
                } else if (!strcmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
                        /* This is the referrer */
@@ -6474,7 +6474,7 @@ static int sip_call(struct ast_channel *ast, const char *dest, int timeout)
 
                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);
@@ -24821,7 +24821,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
 
                /* 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 */
 

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index fa39652..6c7bad9 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -396,6 +396,9 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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
                                 ; for peers and users as well

diff --git a/doc/asterisk.8 b/doc/asterisk.8
index 6cdc7f4..15d7ab7 100644 (file)
--- a/doc/asterisk.8
+++ b/doc/asterisk.8
@@ -16,27 +16,27 @@ asterisk
 \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]
-'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]
+'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]
+[\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]
+[\fB\-v\fR] [\fB\-d\fR] [\fB\-x \fR\fIcommand\fR]
 'in \n(.iu-\nxu
 .ad b
 'hy
@@ -52,12 +52,12 @@ ISDN BRI and many more.
 .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 @@ in the foreground). However running it with \*(T<\fB\-r\fR\*(T> or
 \*(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 @@ has decreased to under \fImemory\fR megabytes.
 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.
+with the \fB\-U\fR or \fB\-G\fR flags.
 .TP 
--G \fIgroup\fR
+\-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 @@ Note: \fBastcanary\fR will run concurrently with
 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).
+console mode (\fB\-c\fR).
 .TP 
--r
+\-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
+\fBasterisk \-vvvgc\fR - Run on controlling terminal
 .PP
-\fBasterisk -rx "core show channels"\fR - Display channels on running server
+\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"

diff --git a/main/features.c b/main/features.c
index cb4abbb..da31288 100644 (file)
--- a/main/features.c
+++ b/main/features.c
@@ -1009,7 +1009,7 @@ int ast_bridge_timelimit(struct ast_channel *chan, struct ast_bridge_config *con
  *
  * 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)
 {