From 0fe1a581e5978c444f67c9fbba6aab7ee1fa0d3e Mon Sep 17 00:00:00 2001 From: Naveen Albert Date: Wed, 1 Jun 2022 00:49:12 +0000 Subject: [PATCH] general: Fix various typos. ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275 --- apps/app_confbridge.c | 2 +- apps/app_dial.c | 4 ++-- apps/app_playback.c | 4 ++-- channels/chan_dahdi.c | 4 ++-- channels/iax2/include/iax2.h | 2 +- channels/sig_analog.c | 14 +++++++------- channels/sig_analog.h | 2 +- configs/samples/iax.conf.sample | 2 +- funcs/func_logic.c | 4 ++-- include/asterisk/test.h | 2 +- main/asterisk.c | 6 +++--- main/bridge.c | 2 +- main/channel.c | 2 +- main/db.c | 6 +++--- res/res_mutestream.c | 2 +- res/res_tonedetect.c | 2 +- 16 files changed, 30 insertions(+), 30 deletions(-) diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c index 9b3ddba836..24dc63e134 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 4c4ebeb5fd..c3892254b8 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -372,7 +372,7 @@ Enables operator services 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 56e74ac5e8..56c2a86682 100644 --- a/apps/app_playback.c +++ b/apps/app_playback.c @@ -73,8 +73,8 @@ 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 + The Playback application answers the channel if no options are specified. + If the file is non-existent it will fail. This application sets the following channel variable upon completion: diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c index 9135937a41..38290b0d75 100644 --- a/channels/chan_dahdi.c +++ b/channels/chan_dahdi.c @@ -238,8 +238,8 @@ DAHDI allows several modifiers to be specified as part of the resource. The general syntax is : Dial(DAHDI/pseudo[/extension]) - Dial(DAHDI/<channel#>[c|r<cadance#>|d][/extension]) - Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension]) + Dial(DAHDI/<channel#>[c|r<cadence#>|d][/extension]) + Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension]) The following modifiers may be used before the channel number: diff --git a/channels/iax2/include/iax2.h b/channels/iax2/include/iax2.h index e9dc967572..0d92674833 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 fb93d5f3d9..bd16d35614 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 488be3662e..7e9acda55c 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 1d1c136edc..5dee369724 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 d267749333..e62ae54c5b 100644 --- a/funcs/func_logic.c +++ b/funcs/func_logic.c @@ -72,10 +72,10 @@ - Check for an expresion. + Check for an expression. - + diff --git a/include/asterisk/test.h b/include/asterisk/test.h index e23aca8df5..78d9788f7e 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 0d6217b605..2d70c53abd 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 289c48bc09..112b621b43 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 8e1c62946b..97ba0f8b06 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 8965014662..2277791cee 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 93c6d0a9d8..a09c83c36c 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 055142bbef..ec5f784242 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 */