[Asterisk-code-review] general: Fix typos. (asterisk[master])
N A
asteriskteam at digium.com
Tue May 31 19:50:18 CDT 2022
N A has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/18601 )
Change subject: general: Fix typos.
......................................................................
general: Fix typos.
Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
---
M apps/app_confbridge.c
M apps/app_dial.c
M channels/chan_dahdi.c
M channels/chan_iax2.c
M channels/iax2/include/iax2.h
M channels/sig_analog.c
M channels/sig_analog.h
M main/asterisk.c
M main/bridge.c
M main/channel.c
M res/res_mutestream.c
M res/res_tonedetect.c
12 files changed, 22 insertions(+), 22 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/01/18601/1
diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
index ae17bca..765e9a4 100644
--- a/apps/app_confbridge.c
+++ b/apps/app_confbridge.c
@@ -1732,7 +1732,7 @@
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 3cf2343..2357c99 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
@@ -1325,7 +1325,7 @@
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) {
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 9135937..38290b0 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/chan_iax2.c b/channels/chan_iax2.c
index 6d76dc5..8e97ef1 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -14310,7 +14310,7 @@
close(com[1]);
close(com[0]);
if (doabort) {
- /* Don't interpret anything, just abort. Not sure what th epoint
+ /* Don't interpret anything, just abort. Not sure what the point
of undeferring dtmf on a hung up channel is but hey whatever */
if (!old && chan)
ast_channel_undefer_dtmf(chan);
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 ea507fe..842b450 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/main/asterisk.c b/main/asterisk.c
index b965a4d..45f3237 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -297,7 +297,7 @@
#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 \
@@ -3554,7 +3554,7 @@
}
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':
@@ -4063,7 +4063,7 @@
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 289c48b..112b621 100644
--- a/main/bridge.c
+++ b/main/bridge.c
@@ -2525,7 +2525,7 @@
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 8e1c629..97ba0f8 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -6106,7 +6106,7 @@
}
/*
- * 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/res/res_mutestream.c b/res/res_mutestream.c
index 8040a3a..df1e148 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 */
--
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
Gerrit-Change-Number: 18601
Gerrit-PatchSet: 1
Gerrit-Owner: N A <mail at interlinked.x10host.com>
Gerrit-MessageType: newchange
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