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general: Fix various typos. 

ASTERISK-30089 #close

Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
This commit is contained in:
Naveen Albert 2022-06-01 00:49:12 +00:00 committed by George Joseph
parent 4cbe12d6d1
commit bcc18ca9f5
16 changed files with 30 additions and 30 deletions
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2
apps/app_confbridge.c
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@ -1738,7 +1738,7 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
struct post_join_action *action;
int max_members_reached = 0;
/* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same */
/* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same time */
ao2_lock(conference_bridges);
ast_debug(1, "Trying to find conference bridge '%s'\n", conference_name);

4
apps/app_dial.c
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@ -372,7 +372,7 @@
</argument>
<para>Enables <emphasis>operator services</emphasis> mode. This option only
works when bridging a DAHDI channel to another DAHDI channel
only. if specified on non-DAHDI interfaces, it will be ignored.
only. If specified on non-DAHDI interfaces, it will be ignored.
When the destination answers (presumably an operator services
station), the originator no longer has control of their line.
They may hang up, but the switch will not release their line
@ -1325,7 +1325,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
if (is_cc_recall) {
ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
}
SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outging channels available\n", ast_channel_name(in));
SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
}
winner = ast_waitfor_n(watchers, pos, to);
AST_LIST_TRAVERSE(out_chans, o, node) {

4
apps/app_playback.c
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@ -73,8 +73,8 @@
</syntax>
<description>
<para>Plays back given filenames (do not put extension of wav/alaw etc).
The playback command answer the channel if no options are specified.
If the file is non-existant it will fail</para>
The Playback application answers the channel if no options are specified.
If the file is non-existent it will fail.</para>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PLAYBACKSTATUS">

4
channels/chan_dahdi.c
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@ -238,8 +238,8 @@
<para>DAHDI allows several modifiers to be specified as part of the resource.</para>
<para>The general syntax is :</para>
<para><literal>Dial(DAHDI/pseudo[/extension])</literal></para>
<para><literal>Dial(DAHDI/&lt;channel#&gt;[c|r&lt;cadance#&gt;|d][/extension])</literal></para>
<para><literal>Dial(DAHDI/(g|G|r|R)&lt;group#(0-63)&gt;[c|r&lt;cadance#&gt;|d][/extension])</literal></para>
<para><literal>Dial(DAHDI/&lt;channel#&gt;[c|r&lt;cadence#&gt;|d][/extension])</literal></para>
<para><literal>Dial(DAHDI/(g|G|r|R)&lt;group#(0-63)&gt;[c|r&lt;cadence#&gt;|d][/extension])</literal></para>
<para>The following modifiers may be used before the channel number:</para>
<enumlist>
<enum name="g">

2
channels/iax2/include/iax2.h
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@ -75,7 +75,7 @@ enum iax_frame_subclass {
IAX_COMMAND_VNAK = 18,
/*! Request status of a dialplan entry */
IAX_COMMAND_DPREQ = 19,
/*! Request status of a dialplan entry */
/*! Status reply of a dialplan entry status request */
IAX_COMMAND_DPREP = 20,
/*! Request a dial on channel brought up TBD */
IAX_COMMAND_DIAL = 21,

14
channels/sig_analog.c
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@ -2235,12 +2235,12 @@ static void *__analog_ss_thread(void *data)
} else if (!strcmp(exten, pickupexten)) {
/* Scan all channels and see if there are any
* ringing channels that have call groups
* that equal this channels pickup group
* that equal this channel's pickup group
*/
if (idx == ANALOG_SUB_REAL) {
/* Switch us from Third call to Call Wait */
if (p->subs[ANALOG_SUB_THREEWAY].owner) {
/* If you make a threeway call and the *8# a call, it should actually
/* If you make a threeway call and then *8# a call, it should actually
look like a callwait */
analog_alloc_sub(p, ANALOG_SUB_CALLWAIT);
analog_swap_subs(p, ANALOG_SUB_CALLWAIT, ANALOG_SUB_THREEWAY);
@ -2808,7 +2808,7 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
switch (res) {
case ANALOG_EVENT_EC_DISABLED:
ast_verb(3, "Channel %d echo canceler disabled due to CED detection\n", p->channel);
ast_verb(3, "Channel %d echo canceller disabled due to CED detection\n", p->channel);
analog_set_echocanceller(p, 0);
break;
#ifdef HAVE_DAHDI_ECHOCANCEL_FAX_MODE
@ -2819,10 +2819,10 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
ast_verb(3, "Channel %d detected a CED tone from the network.\n", p->channel);
break;
case ANALOG_EVENT_EC_NLP_DISABLED:
ast_verb(3, "Channel %d echo canceler disabled its NLP.\n", p->channel);
ast_verb(3, "Channel %d echo canceller disabled its NLP.\n", p->channel);
break;
case ANALOG_EVENT_EC_NLP_ENABLED:
ast_verb(3, "Channel %d echo canceler enabled its NLP.\n", p->channel);
ast_verb(3, "Channel %d echo canceller enabled its NLP.\n", p->channel);
break;
#endif
case ANALOG_EVENT_PULSE_START:
@ -2907,14 +2907,14 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
analog_lock_sub_owner(p, ANALOG_SUB_CALLWAIT);
if (!p->subs[ANALOG_SUB_CALLWAIT].owner) {
/*
* The call waiting call dissappeared.
* The call waiting call disappeared.
* This is now a normal hangup.
*/
analog_set_echocanceller(p, 0);
return NULL;
}
/* There's a call waiting call, so ring the phone, but make it unowned in the mean time */
/* There's a call waiting call, so ring the phone, but make it unowned in the meantime */
analog_swap_subs(p, ANALOG_SUB_CALLWAIT, ANALOG_SUB_REAL);
ast_verb(3, "Channel %d still has (callwait) call, ringing phone\n", p->channel);
analog_unalloc_sub(p, ANALOG_SUB_CALLWAIT);

2
channels/sig_analog.h
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@ -266,7 +266,7 @@ struct analog_pvt {
enum analog_sigtype sig;
/* To contain the private structure passed into the channel callbacks */
void *chan_pvt;
/* All members after this are giong to be transient, and most will probably change */
/* All members after this are going to be transient, and most will probably change */
struct ast_channel *owner; /*!< Our current active owner (if applicable) */
struct analog_subchannel subs[3]; /*!< Sub-channels */

2
configs/samples/iax.conf.sample
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@ -386,7 +386,7 @@ autokill=yes
; IAX2 clients which request it. This has only been used for the IAXy,
; and it has been recently proven that this firmware distribution method
; can be used as a source of traffic amplification attacks. Also, the
; IAXy firmware has not been updated for at least 18 months, so unless
; IAXy firmware has not been updated since at least 2012, so unless
; you are provisioning IAXys in a secure network, we recommend that you
; leave this option to the default, off.
;

4
funcs/func_logic.c
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@ -72,10 +72,10 @@
</function>
<function name="IF" language="en_US">
<synopsis>
Check for an expresion.
Check for an expression.
</synopsis>
<syntax argsep="?">
<parameter name="expresion" required="true" />
<parameter name="expression" required="true" />
<parameter name="retvalue" argsep=":" required="true">
<argument name="true" />
<argument name="false" />

2
include/asterisk/test.h
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@ -108,7 +108,7 @@
\code
'test show registered all' will show every registered test.
'test execute all' will execute every registered test.
'test show results all' will show detailed results for ever executed test
'test show results all' will show detailed results for every executed test
'test generate results xml' will generate a test report in xml format
'test generate results txt' will generate a test report in txt format
\endcode

6
main/asterisk.c
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@ -297,7 +297,7 @@ int daemon(int, int); /* defined in libresolv of all places */
#define NUM_MSGS 64
/*! Displayed copyright tag */
#define COPYRIGHT_TAG "Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others."
#define COPYRIGHT_TAG "Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others."
/*! \brief Welcome message when starting a CLI interface */
#define WELCOME_MESSAGE \
@ -3571,7 +3571,7 @@ int main(int argc, char *argv[])
}
ast_mainpid = getpid();
/* Process command-line options that effect asterisk.conf load. */
/* Process command-line options that affect asterisk.conf load. */
while ((c = getopt(argc, argv, getopt_settings)) != -1) {
switch (c) {
case 'X':
@ -4082,7 +4082,7 @@ static void asterisk_daemon(int isroot, const char *runuser, const char *rungrou
load_astmm_phase_1();
/* Check whether high prio was succesfully set by us or some
/* Check whether high prio was successfully set by us or some
* other incantation. */
if (has_priority()) {
ast_set_flag(&ast_options, AST_OPT_FLAG_HIGH_PRIORITY);

2
main/bridge.c
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@ -2525,7 +2525,7 @@ int ast_bridge_add_channel(struct ast_bridge *bridge, struct ast_channel *chan,
if (ast_bridge_impart(bridge, yanked_chan, NULL, features,
AST_BRIDGE_IMPART_CHAN_INDEPENDENT)) {
/* It is possible for us to yank a channel and have some other
* thread start a PBX on the channl after we yanked it. In particular,
* thread start a PBX on the channel after we yanked it. In particular,
* this can theoretically happen on the ;2 of a Local channel if we
* yank it prior to the ;1 being answered. Make sure that it isn't
* executing a PBX before hanging it up.

2
main/channel.c
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@ -6106,7 +6106,7 @@ struct ast_channel *__ast_request_and_dial(const char *type, struct ast_format_c
}
/*
* I seems strange to set the CallerID on an outgoing call leg
* It seems strange to set the CallerID on an outgoing call leg
* to whom we are calling, but this function's callers are doing
* various Originate methods. This call leg goes to the local
* user. Once the local user answers, the dialplan needs to be

6
main/db.c
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@ -507,7 +507,7 @@ int ast_db_deltree(const char *family, const char *keytree)
ast_mutex_lock(&dblock);
if (!ast_strlen_zero(prefix) && (sqlite3_bind_text(stmt, 1, prefix, -1, SQLITE_STATIC) != SQLITE_OK)) {
ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));
ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));
res = -1;
} else if (sqlite3_step(stmt) != SQLITE_DONE) {
ast_log(LOG_WARNING, "Couldn't execute stmt: %s\n", sqlite3_errmsg(astdb));
@ -791,7 +791,7 @@ static char *handle_cli_database_show(struct ast_cli_entry *e, int cmd, struct a
ast_mutex_lock(&dblock);
if (!ast_strlen_zero(prefix) && (sqlite3_bind_text(stmt, 1, prefix, -1, SQLITE_STATIC) != SQLITE_OK)) {
ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));
ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));
sqlite3_reset(stmt);
ast_mutex_unlock(&dblock);
return NULL;
@ -839,7 +839,7 @@ static char *handle_cli_database_showkey(struct ast_cli_entry *e, int cmd, struc
ast_mutex_lock(&dblock);
if (!ast_strlen_zero(a->argv[2]) && (sqlite3_bind_text(showkey_stmt, 1, a->argv[2], -1, SQLITE_STATIC) != SQLITE_OK)) {
ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", a->argv[2], sqlite3_errmsg(astdb));
ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", a->argv[2], sqlite3_errmsg(astdb));
sqlite3_reset(showkey_stmt);
ast_mutex_unlock(&dblock);
return NULL;

2
res/res_mutestream.c
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@ -26,7 +26,7 @@
*
* \note This module only handles audio streams today, but can easily be appended to also
* zero out text streams if there's an application for it.
* When we know and understands what happens if we zero out video, we can do that too.
* When we know and understand what happens if we zero out video, we can do that too.
*/
/*** MODULEINFO

2
res/res_tonedetect.c
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@ -902,7 +902,7 @@ static int scan_exec(struct ast_channel *chan, const char *data)
}
ast_dsp_set_features(dsp, features);
/* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */
ast_dsp_set_freqmode(dsp, 2225, 400, 16, 0); /* this needs to be pretty short, or the progress tones code will thing this is voice */
ast_dsp_set_freqmode(dsp, 2225, 400, 16, 0); /* this needs to be pretty short, or the progress tones code will think this is voice */
if (fax) { /* fax detect uses same tone detect internals as modem and causes things to not work as intended, so use a separate DSP if needed. */
ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */