<p>George Joseph <strong>submitted</strong> this change.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/18601">View Change</a></p><div style="white-space:pre-wrap">Approvals:
Joshua Colp: Looks good to me, but someone else must approve
Kevin Harwell: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved; Approved for Submit
</div><pre style="font-family: monospace,monospace; white-space: pre-wrap;">general: Fix various typos.<br><br>ASTERISK-30089 #close<br><br>Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275<br>---<br>M apps/app_confbridge.c<br>M apps/app_dial.c<br>M apps/app_playback.c<br>M channels/chan_dahdi.c<br>M channels/iax2/include/iax2.h<br>M channels/sig_analog.c<br>M channels/sig_analog.h<br>M configs/samples/iax.conf.sample<br>M funcs/func_logic.c<br>M include/asterisk/test.h<br>M main/asterisk.c<br>M main/bridge.c<br>M main/channel.c<br>M main/db.c<br>M res/res_mutestream.c<br>M res/res_tonedetect.c<br>16 files changed, 30 insertions(+), 30 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c</span><br><span>index 9b3ddba..24dc63e 100644</span><br><span>--- a/apps/app_confbridge.c</span><br><span>+++ b/apps/app_confbridge.c</span><br><span>@@ -1738,7 +1738,7 @@</span><br><span> struct post_join_action *action;</span><br><span> int max_members_reached = 0;</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- /* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same time */</span><br><span> ao2_lock(conference_bridges);</span><br><span> </span><br><span> ast_debug(1, "Trying to find conference bridge '%s'\n", conference_name);</span><br><span>diff --git a/apps/app_dial.c b/apps/app_dial.c</span><br><span>index 4c4ebeb..c389225 100644</span><br><span>--- a/apps/app_dial.c</span><br><span>+++ b/apps/app_dial.c</span><br><span>@@ -372,7 +372,7 @@</span><br><span> </argument></span><br><span> <para>Enables <emphasis>operator services</emphasis> mode. This option only</span><br><span> works when bridging a DAHDI channel to another DAHDI channel</span><br><span style="color: hsl(0, 100%, 40%);">- only. if specified on non-DAHDI interfaces, it will be ignored.</span><br><span style="color: hsl(120, 100%, 40%);">+ only. If specified on non-DAHDI interfaces, it will be ignored.</span><br><span> When the destination answers (presumably an operator services</span><br><span> station), the originator no longer has control of their line.</span><br><span> They may hang up, but the switch will not release their line</span><br><span>@@ -1325,7 +1325,7 @@</span><br><span> if (is_cc_recall) {</span><br><span> ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");</span><br><span> }</span><br><span style="color: hsl(0, 100%, 40%);">- SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outging channels available\n", ast_channel_name(in));</span><br><span style="color: hsl(120, 100%, 40%);">+ SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));</span><br><span> }</span><br><span> winner = ast_waitfor_n(watchers, pos, to);</span><br><span> AST_LIST_TRAVERSE(out_chans, o, node) {</span><br><span>diff --git a/apps/app_playback.c b/apps/app_playback.c</span><br><span>index 56e74ac..56c2a86 100644</span><br><span>--- a/apps/app_playback.c</span><br><span>+++ b/apps/app_playback.c</span><br><span>@@ -73,8 +73,8 @@</span><br><span> </syntax></span><br><span> <description></span><br><span> <para>Plays back given filenames (do not put extension of wav/alaw etc).</span><br><span style="color: hsl(0, 100%, 40%);">- The playback command answer the channel if no options are specified.</span><br><span style="color: hsl(0, 100%, 40%);">- If the file is non-existant it will fail</para></span><br><span style="color: hsl(120, 100%, 40%);">+ The Playback application answers the channel if no options are specified.</span><br><span style="color: hsl(120, 100%, 40%);">+ If the file is non-existent it will fail.</para></span><br><span> <para>This application sets the following channel variable upon completion:</para></span><br><span> <variablelist></span><br><span> <variable name="PLAYBACKSTATUS"></span><br><span>diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c</span><br><span>index 9135937..38290b0 100644</span><br><span>--- a/channels/chan_dahdi.c</span><br><span>+++ b/channels/chan_dahdi.c</span><br><span>@@ -238,8 +238,8 @@</span><br><span> <para>DAHDI allows several modifiers to be specified as part of the resource.</para></span><br><span> <para>The general syntax is :</para></span><br><span> <para><literal>Dial(DAHDI/pseudo[/extension])</literal></para></span><br><span style="color: hsl(0, 100%, 40%);">- <para><literal>Dial(DAHDI/<channel#>[c|r<cadance#>|d][/extension])</literal></para></span><br><span style="color: hsl(0, 100%, 40%);">- <para><literal>Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])</literal></para></span><br><span style="color: hsl(120, 100%, 40%);">+ <para><literal>Dial(DAHDI/<channel#>[c|r<cadence#>|d][/extension])</literal></para></span><br><span style="color: hsl(120, 100%, 40%);">+ <para><literal>Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadence#>|d][/extension])</literal></para></span><br><span> <para>The following modifiers may be used before the channel number:</para></span><br><span> <enumlist></span><br><span> <enum name="g"></span><br><span>diff --git a/channels/iax2/include/iax2.h b/channels/iax2/include/iax2.h</span><br><span>index e9dc967..0d92674 100644</span><br><span>--- a/channels/iax2/include/iax2.h</span><br><span>+++ b/channels/iax2/include/iax2.h</span><br><span>@@ -75,7 +75,7 @@</span><br><span> IAX_COMMAND_VNAK = 18,</span><br><span> /*! Request status of a dialplan entry */</span><br><span> IAX_COMMAND_DPREQ = 19,</span><br><span style="color: hsl(0, 100%, 40%);">- /*! Request status of a dialplan entry */</span><br><span style="color: hsl(120, 100%, 40%);">+ /*! Status reply of a dialplan entry status request */</span><br><span> IAX_COMMAND_DPREP = 20,</span><br><span> /*! Request a dial on channel brought up TBD */</span><br><span> IAX_COMMAND_DIAL = 21,</span><br><span>diff --git a/channels/sig_analog.c b/channels/sig_analog.c</span><br><span>index fb93d5f..bd16d35 100644</span><br><span>--- a/channels/sig_analog.c</span><br><span>+++ b/channels/sig_analog.c</span><br><span>@@ -2235,12 +2235,12 @@</span><br><span> } else if (!strcmp(exten, pickupexten)) {</span><br><span> /* Scan all channels and see if there are any</span><br><span> * ringing channels that have call groups</span><br><span style="color: hsl(0, 100%, 40%);">- * that equal this channels pickup group</span><br><span style="color: hsl(120, 100%, 40%);">+ * that equal this channel's pickup group</span><br><span> */</span><br><span> if (idx == ANALOG_SUB_REAL) {</span><br><span> /* Switch us from Third call to Call Wait */</span><br><span> if (p->subs[ANALOG_SUB_THREEWAY].owner) {</span><br><span style="color: hsl(0, 100%, 40%);">- /* If you make a threeway call and the *8# a call, it should actually</span><br><span style="color: hsl(120, 100%, 40%);">+ /* If you make a threeway call and then *8# a call, it should actually</span><br><span> look like a callwait */</span><br><span> analog_alloc_sub(p, ANALOG_SUB_CALLWAIT);</span><br><span> analog_swap_subs(p, ANALOG_SUB_CALLWAIT, ANALOG_SUB_THREEWAY);</span><br><span>@@ -2808,7 +2808,7 @@</span><br><span> </span><br><span> switch (res) {</span><br><span> case ANALOG_EVENT_EC_DISABLED:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_verb(3, "Channel %d echo canceler disabled due to CED detection\n", p->channel);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_verb(3, "Channel %d echo canceller disabled due to CED detection\n", p->channel);</span><br><span> analog_set_echocanceller(p, 0);</span><br><span> break;</span><br><span> #ifdef HAVE_DAHDI_ECHOCANCEL_FAX_MODE</span><br><span>@@ -2819,10 +2819,10 @@</span><br><span> ast_verb(3, "Channel %d detected a CED tone from the network.\n", p->channel);</span><br><span> break;</span><br><span> case ANALOG_EVENT_EC_NLP_DISABLED:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_verb(3, "Channel %d echo canceler disabled its NLP.\n", p->channel);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_verb(3, "Channel %d echo canceller disabled its NLP.\n", p->channel);</span><br><span> break;</span><br><span> case ANALOG_EVENT_EC_NLP_ENABLED:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_verb(3, "Channel %d echo canceler enabled its NLP.\n", p->channel);</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_verb(3, "Channel %d echo canceller enabled its NLP.\n", p->channel);</span><br><span> break;</span><br><span> #endif</span><br><span> case ANALOG_EVENT_PULSE_START:</span><br><span>@@ -2907,14 +2907,14 @@</span><br><span> analog_lock_sub_owner(p, ANALOG_SUB_CALLWAIT);</span><br><span> if (!p->subs[ANALOG_SUB_CALLWAIT].owner) {</span><br><span> /*</span><br><span style="color: hsl(0, 100%, 40%);">- * The call waiting call dissappeared.</span><br><span style="color: hsl(120, 100%, 40%);">+ * The call waiting call disappeared.</span><br><span> * This is now a normal hangup.</span><br><span> */</span><br><span> analog_set_echocanceller(p, 0);</span><br><span> return NULL;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- /* There's a call waiting call, so ring the phone, but make it unowned in the mean time */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* There's a call waiting call, so ring the phone, but make it unowned in the meantime */</span><br><span> analog_swap_subs(p, ANALOG_SUB_CALLWAIT, ANALOG_SUB_REAL);</span><br><span> ast_verb(3, "Channel %d still has (callwait) call, ringing phone\n", p->channel);</span><br><span> analog_unalloc_sub(p, ANALOG_SUB_CALLWAIT);</span><br><span>diff --git a/channels/sig_analog.h b/channels/sig_analog.h</span><br><span>index 488be36..7e9acda 100644</span><br><span>--- a/channels/sig_analog.h</span><br><span>+++ b/channels/sig_analog.h</span><br><span>@@ -266,7 +266,7 @@</span><br><span> enum analog_sigtype sig;</span><br><span> /* To contain the private structure passed into the channel callbacks */</span><br><span> void *chan_pvt;</span><br><span style="color: hsl(0, 100%, 40%);">- /* All members after this are giong to be transient, and most will probably change */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* All members after this are going to be transient, and most will probably change */</span><br><span> struct ast_channel *owner; /*!< Our current active owner (if applicable) */</span><br><span> </span><br><span> struct analog_subchannel subs[3]; /*!< Sub-channels */</span><br><span>diff --git a/configs/samples/iax.conf.sample b/configs/samples/iax.conf.sample</span><br><span>index 1d1c136..5dee369 100644</span><br><span>--- a/configs/samples/iax.conf.sample</span><br><span>+++ b/configs/samples/iax.conf.sample</span><br><span>@@ -386,7 +386,7 @@</span><br><span> ; IAX2 clients which request it. This has only been used for the IAXy,</span><br><span> ; and it has been recently proven that this firmware distribution method</span><br><span> ; can be used as a source of traffic amplification attacks. Also, the</span><br><span style="color: hsl(0, 100%, 40%);">-; IAXy firmware has not been updated for at least 18 months, so unless</span><br><span style="color: hsl(120, 100%, 40%);">+; IAXy firmware has not been updated since at least 2012, so unless</span><br><span> ; you are provisioning IAXys in a secure network, we recommend that you</span><br><span> ; leave this option to the default, off.</span><br><span> ;</span><br><span>diff --git a/funcs/func_logic.c b/funcs/func_logic.c</span><br><span>index d267749..e62ae54 100644</span><br><span>--- a/funcs/func_logic.c</span><br><span>+++ b/funcs/func_logic.c</span><br><span>@@ -72,10 +72,10 @@</span><br><span> </function></span><br><span> <function name="IF" language="en_US"></span><br><span> <synopsis></span><br><span style="color: hsl(0, 100%, 40%);">- Check for an expresion.</span><br><span style="color: hsl(120, 100%, 40%);">+ Check for an expression.</span><br><span> </synopsis></span><br><span> <syntax argsep="?"></span><br><span style="color: hsl(0, 100%, 40%);">- <parameter name="expresion" required="true" /></span><br><span style="color: hsl(120, 100%, 40%);">+ <parameter name="expression" required="true" /></span><br><span> <parameter name="retvalue" argsep=":" required="true"></span><br><span> <argument name="true" /></span><br><span> <argument name="false" /></span><br><span>diff --git a/include/asterisk/test.h b/include/asterisk/test.h</span><br><span>index e23aca8..78d9788 100644</span><br><span>--- a/include/asterisk/test.h</span><br><span>+++ b/include/asterisk/test.h</span><br><span>@@ -108,7 +108,7 @@</span><br><span> \code</span><br><span> 'test show registered all' will show every registered test.</span><br><span> 'test execute all' will execute every registered test.</span><br><span style="color: hsl(0, 100%, 40%);">- 'test show results all' will show detailed results for ever executed test</span><br><span style="color: hsl(120, 100%, 40%);">+ 'test show results all' will show detailed results for every executed test</span><br><span> 'test generate results xml' will generate a test report in xml format</span><br><span> 'test generate results txt' will generate a test report in txt format</span><br><span> \endcode</span><br><span>diff --git a/main/asterisk.c b/main/asterisk.c</span><br><span>index 0d6217b..2d70c53 100644</span><br><span>--- a/main/asterisk.c</span><br><span>+++ b/main/asterisk.c</span><br><span>@@ -297,7 +297,7 @@</span><br><span> #define NUM_MSGS 64</span><br><span> </span><br><span> /*! Displayed copyright tag */</span><br><span style="color: hsl(0, 100%, 40%);">-#define COPYRIGHT_TAG "Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others."</span><br><span style="color: hsl(120, 100%, 40%);">+#define COPYRIGHT_TAG "Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others."</span><br><span> </span><br><span> /*! \brief Welcome message when starting a CLI interface */</span><br><span> #define WELCOME_MESSAGE \</span><br><span>@@ -3571,7 +3571,7 @@</span><br><span> }</span><br><span> ast_mainpid = getpid();</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- /* Process command-line options that effect asterisk.conf load. */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Process command-line options that affect asterisk.conf load. */</span><br><span> while ((c = getopt(argc, argv, getopt_settings)) != -1) {</span><br><span> switch (c) {</span><br><span> case 'X':</span><br><span>@@ -4082,7 +4082,7 @@</span><br><span> </span><br><span> load_astmm_phase_1();</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- /* Check whether high prio was succesfully set by us or some</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Check whether high prio was successfully set by us or some</span><br><span> * other incantation. */</span><br><span> if (has_priority()) {</span><br><span> ast_set_flag(&ast_options, AST_OPT_FLAG_HIGH_PRIORITY);</span><br><span>diff --git a/main/bridge.c b/main/bridge.c</span><br><span>index 289c48b..112b621 100644</span><br><span>--- a/main/bridge.c</span><br><span>+++ b/main/bridge.c</span><br><span>@@ -2525,7 +2525,7 @@</span><br><span> if (ast_bridge_impart(bridge, yanked_chan, NULL, features,</span><br><span> AST_BRIDGE_IMPART_CHAN_INDEPENDENT)) {</span><br><span> /* It is possible for us to yank a channel and have some other</span><br><span style="color: hsl(0, 100%, 40%);">- * thread start a PBX on the channl after we yanked it. In particular,</span><br><span style="color: hsl(120, 100%, 40%);">+ * thread start a PBX on the channel after we yanked it. In particular,</span><br><span> * this can theoretically happen on the ;2 of a Local channel if we</span><br><span> * yank it prior to the ;1 being answered. Make sure that it isn't</span><br><span> * executing a PBX before hanging it up.</span><br><span>diff --git a/main/channel.c b/main/channel.c</span><br><span>index 8e1c629..97ba0f8 100644</span><br><span>--- a/main/channel.c</span><br><span>+++ b/main/channel.c</span><br><span>@@ -6106,7 +6106,7 @@</span><br><span> }</span><br><span> </span><br><span> /*</span><br><span style="color: hsl(0, 100%, 40%);">- * I seems strange to set the CallerID on an outgoing call leg</span><br><span style="color: hsl(120, 100%, 40%);">+ * It seems strange to set the CallerID on an outgoing call leg</span><br><span> * to whom we are calling, but this function's callers are doing</span><br><span> * various Originate methods. This call leg goes to the local</span><br><span> * user. Once the local user answers, the dialplan needs to be</span><br><span>diff --git a/main/db.c b/main/db.c</span><br><span>index 8965014..2277791 100644</span><br><span>--- a/main/db.c</span><br><span>+++ b/main/db.c</span><br><span>@@ -507,7 +507,7 @@</span><br><span> </span><br><span> ast_mutex_lock(&dblock);</span><br><span> if (!ast_strlen_zero(prefix) && (sqlite3_bind_text(stmt, 1, prefix, -1, SQLITE_STATIC) != SQLITE_OK)) {</span><br><span style="color: hsl(0, 100%, 40%);">- ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));</span><br><span> res = -1;</span><br><span> } else if (sqlite3_step(stmt) != SQLITE_DONE) {</span><br><span> ast_log(LOG_WARNING, "Couldn't execute stmt: %s\n", sqlite3_errmsg(astdb));</span><br><span>@@ -791,7 +791,7 @@</span><br><span> </span><br><span> ast_mutex_lock(&dblock);</span><br><span> if (!ast_strlen_zero(prefix) && (sqlite3_bind_text(stmt, 1, prefix, -1, SQLITE_STATIC) != SQLITE_OK)) {</span><br><span style="color: hsl(0, 100%, 40%);">- ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", prefix, sqlite3_errmsg(astdb));</span><br><span> sqlite3_reset(stmt);</span><br><span> ast_mutex_unlock(&dblock);</span><br><span> return NULL;</span><br><span>@@ -839,7 +839,7 @@</span><br><span> </span><br><span> ast_mutex_lock(&dblock);</span><br><span> if (!ast_strlen_zero(a->argv[2]) && (sqlite3_bind_text(showkey_stmt, 1, a->argv[2], -1, SQLITE_STATIC) != SQLITE_OK)) {</span><br><span style="color: hsl(0, 100%, 40%);">- ast_log(LOG_WARNING, "Could bind %s to stmt: %s\n", a->argv[2], sqlite3_errmsg(astdb));</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_log(LOG_WARNING, "Couldn't bind %s to stmt: %s\n", a->argv[2], sqlite3_errmsg(astdb));</span><br><span> sqlite3_reset(showkey_stmt);</span><br><span> ast_mutex_unlock(&dblock);</span><br><span> return NULL;</span><br><span>diff --git a/res/res_mutestream.c b/res/res_mutestream.c</span><br><span>index 93c6d0a..a09c83c 100644</span><br><span>--- a/res/res_mutestream.c</span><br><span>+++ b/res/res_mutestream.c</span><br><span>@@ -26,7 +26,7 @@</span><br><span> *</span><br><span> * \note This module only handles audio streams today, but can easily be appended to also</span><br><span> * zero out text streams if there's an application for it.</span><br><span style="color: hsl(0, 100%, 40%);">- * When we know and understands what happens if we zero out video, we can do that too.</span><br><span style="color: hsl(120, 100%, 40%);">+ * When we know and understand what happens if we zero out video, we can do that too.</span><br><span> */</span><br><span> </span><br><span> /*** MODULEINFO</span><br><span>diff --git a/res/res_tonedetect.c b/res/res_tonedetect.c</span><br><span>index 055142b..ec5f784 100644</span><br><span>--- a/res/res_tonedetect.c</span><br><span>+++ b/res/res_tonedetect.c</span><br><span>@@ -902,7 +902,7 @@</span><br><span> }</span><br><span> ast_dsp_set_features(dsp, features);</span><br><span> /* all modems begin negotiating with Bell 103. An answering modem just sends mark tone, or 2225 Hz */</span><br><span style="color: hsl(0, 100%, 40%);">- 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 */</span><br><span style="color: hsl(120, 100%, 40%);">+ 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 */</span><br><span> </span><br><span> 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. */</span><br><span> ast_dsp_set_features(dsp2, DSP_FEATURE_FAX_DETECT); /* fax tone */</span><br><span></span><br></pre><div style="white-space:pre-wrap"></div><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/18601">change 18601</a>. To unsubscribe, or for help writing mail filters, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/c/asterisk/+/18601"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275 </div>
<div style="display:none"> Gerrit-Change-Number: 18601 </div>
<div style="display:none"> Gerrit-PatchSet: 8 </div>
<div style="display:none"> Gerrit-Owner: N A <mail@interlinked.x10host.com> </div>
<div style="display:none"> Gerrit-Reviewer: Friendly Automation </div>
<div style="display:none"> Gerrit-Reviewer: George Joseph <gjoseph@digium.com> </div>
<div style="display:none"> Gerrit-Reviewer: Joshua Colp <jcolp@sangoma.com> </div>
<div style="display:none"> Gerrit-Reviewer: Kevin Harwell <kharwell@digium.com> </div>
<div style="display:none"> Gerrit-MessageType: merged </div>