[Asterisk-code-review] general: Fix various typos. (asterisk[16])
George Joseph
asteriskteam at digium.com
Tue Jul 12 07:45:03 CDT 2022
George Joseph has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/18756 )
Change subject: general: Fix various typos.
......................................................................
general: Fix various typos.
ASTERISK-30089 #close
Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
---
M apps/app_confbridge.c
M apps/app_dial.c
M apps/app_playback.c
M channels/chan_dahdi.c
M channels/iax2/include/iax2.h
M channels/sig_analog.c
M channels/sig_analog.h
M configs/samples/iax.conf.sample
M funcs/func_logic.c
M include/asterisk/test.h
M main/asterisk.c
M main/bridge.c
M main/channel.c
M main/db.c
M res/res_mutestream.c
M res/res_tonedetect.c
16 files changed, 28 insertions(+), 28 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
Benjamin Keith Ford: Looks good to me, but someone else must approve
Michael Bradeen: Looks good to me, approved
George Joseph: Approved for Submit
diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
index 4052771..492649c 100644
--- a/apps/app_confbridge.c
+++ b/apps/app_confbridge.c
@@ -1738,7 +1738,7 @@
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);
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 031e26f..7fd6f40 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -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
diff --git a/apps/app_playback.c b/apps/app_playback.c
index 56e74ac..56c2a86 100644
--- a/apps/app_playback.c
+++ b/apps/app_playback.c
@@ -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">
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 506c7a9..3564040 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -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/<channel#>[c|r<cadance#>|d][/extension])</literal></para>
- <para><literal>Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])</literal></para>
+ <para><literal>Dial(DAHDI/<channel#>[c|r<cadence#>|d][/extension])</literal></para>
+ <para><literal>Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension])</literal></para>
<para>The following modifiers may be used before the channel number:</para>
<enumlist>
<enum name="g">
diff --git a/channels/iax2/include/iax2.h b/channels/iax2/include/iax2.h
index e9dc967..0d92674 100644
--- a/channels/iax2/include/iax2.h
+++ b/channels/iax2/include/iax2.h
@@ -75,7 +75,7 @@
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,
diff --git a/channels/sig_analog.c b/channels/sig_analog.c
index fb93d5f..bd16d35 100644
--- a/channels/sig_analog.c
+++ b/channels/sig_analog.c
@@ -2235,12 +2235,12 @@
} 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 @@
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 @@
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 @@
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);
diff --git a/channels/sig_analog.h b/channels/sig_analog.h
index 488be36..7e9acda 100644
--- a/channels/sig_analog.h
+++ b/channels/sig_analog.h
@@ -266,7 +266,7 @@
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 */
diff --git a/configs/samples/iax.conf.sample b/configs/samples/iax.conf.sample
index 1d1c136..5dee369 100644
--- a/configs/samples/iax.conf.sample
+++ b/configs/samples/iax.conf.sample
@@ -386,7 +386,7 @@
; 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.
;
diff --git a/funcs/func_logic.c b/funcs/func_logic.c
index d267749..e62ae54 100644
--- a/funcs/func_logic.c
+++ b/funcs/func_logic.c
@@ -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" />
diff --git a/include/asterisk/test.h b/include/asterisk/test.h
index e23aca8..78d9788 100644
--- a/include/asterisk/test.h
+++ b/include/asterisk/test.h
@@ -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
diff --git a/main/asterisk.c b/main/asterisk.c
index 8a98ea1..9bc4c85 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -3568,7 +3568,7 @@
}
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':
@@ -4079,7 +4079,7 @@
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);
diff --git a/main/bridge.c b/main/bridge.c
index 05eeef4..283ea79 100644
--- a/main/bridge.c
+++ b/main/bridge.c
@@ -2544,7 +2544,7 @@
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.
diff --git a/main/channel.c b/main/channel.c
index d736622..c051760 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -6118,7 +6118,7 @@
}
/*
- * 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
diff --git a/main/db.c b/main/db.c
index 8965014..2277791 100644
--- a/main/db.c
+++ b/main/db.c
@@ -507,7 +507,7 @@
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 @@
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 @@
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;
diff --git a/res/res_mutestream.c b/res/res_mutestream.c
index 93c6d0a..a09c83c 100644
--- a/res/res_mutestream.c
+++ b/res/res_mutestream.c
@@ -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
diff --git a/res/res_tonedetect.c b/res/res_tonedetect.c
index 055142b..ec5f784 100644
--- a/res/res_tonedetect.c
+++ b/res/res_tonedetect.c
@@ -902,7 +902,7 @@
}
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 */
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/18756
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
Gerrit-Change-Number: 18756
Gerrit-PatchSet: 4
Gerrit-Owner: N A <mail at interlinked.x10host.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Michael Bradeen <mbradeen at sangoma.com>
Gerrit-MessageType: merged
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20220712/004181a2/attachment-0001.html>
More information about the asterisk-code-review
mailing list