diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
index 9b3ddba836a7adbfdf57ad11e688557cb6ebf6c4..24dc63e1341fc895709d6ecee70cec616e564e2e 100644
--- a/apps/app_confbridge.c
+++ b/apps/app_confbridge.c
@@ -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);
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 4c4ebeb5fdf22088f3a7e2125613f4dbd5192258..c3892254b8bd24d2dfe52ff5909f08a524832881 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 @@ 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) {
diff --git a/apps/app_playback.c b/apps/app_playback.c
index 56e74ac5e89c253c85675730e83275639595fc49..56c2a86682fef2d737466f6f84f0400a935de8f3 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 9135937a41b5de8769b2a74a8132ce69928f31fc..38290b0d75a519be37ffeea53faa1a8095e05008 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 e9dc96757238dd49981bd2ebf8f0ff1994fbbbef..0d92674833262c9c80c4f2dad90b13d23599a262 100644
--- a/channels/iax2/include/iax2.h
+++ b/channels/iax2/include/iax2.h
@@ -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,
diff --git a/channels/sig_analog.c b/channels/sig_analog.c
index fb93d5f3d9798799a203952e1e30f6b982eb37a7..bd16d3561475b78644e7f7c7c6fb3d7cd3d7da60 100644
--- a/channels/sig_analog.c
+++ b/channels/sig_analog.c
@@ -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);
diff --git a/channels/sig_analog.h b/channels/sig_analog.h
index 488be3662e9ce275c683aba2a278112fb1853ad4..7e9acda55cac35c5ed5819d5c10b4b08c98a17ef 100644
--- a/channels/sig_analog.h
+++ b/channels/sig_analog.h
@@ -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 */
diff --git a/configs/samples/iax.conf.sample b/configs/samples/iax.conf.sample
index 1d1c136edcd8d6ae37816969fe4c1655b2778660..5dee369724dd1fea4e00ada1aa0e754b8304d3f3 100644
--- a/configs/samples/iax.conf.sample
+++ b/configs/samples/iax.conf.sample
@@ -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.
;
diff --git a/funcs/func_logic.c b/funcs/func_logic.c
index d2677493334a305e3c8670c6a3809e76ef93100f..e62ae54c5bac6311e9cd2138aef1b54c471eb4bf 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 e23aca8df55288e519c5eb4982ca5784e7fdb9a8..78d9788f7ebd4686ce3f94bd255a2bfec0c893bb 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 0d6217b6050283ec525df2616979182900d8c134..2d70c53abd1fd53c31964655e863793058b9952f 100644
--- a/main/asterisk.c
+++ b/main/asterisk.c
@@ -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);
diff --git a/main/bridge.c b/main/bridge.c
index 289c48bc09343fd1dd9e758374f8a2c6dc53c267..112b621b43b5bdd15dd52d74e0344c8973c5449e 100644
--- a/main/bridge.c
+++ b/main/bridge.c
@@ -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.
diff --git a/main/channel.c b/main/channel.c
index 8e1c62946be6e982e76517d46ba5359343f7b1f2..97ba0f8b060ce59ce036415295fac3628914490a 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -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
diff --git a/main/db.c b/main/db.c
index 89650146625b18b2c1d3ad8fd67801682b9f7ca4..2277791ceefac96dd4046e6bed02efbfc6b455cb 100644
--- a/main/db.c
+++ b/main/db.c
@@ -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;
diff --git a/res/res_mutestream.c b/res/res_mutestream.c
index 93c6d0a9d8f79adfbe82f563e4d6417d744c3b37..a09c83c36c9db494f73b719843399c42765a2982 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 055142bbef32b906d344a5ea538f6340deda5cc5..ec5f7842423d5cd971f23bcf33be92443bf453c5 100644
--- a/res/res_tonedetect.c
+++ b/res/res_tonedetect.c
@@ -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 */