<p>N A has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/18601">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">general: Fix typos.<br><br>Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275<br>---<br>M apps/app_confbridge.c<br>M apps/app_dial.c<br>M channels/chan_dahdi.c<br>M channels/chan_iax2.c<br>M channels/iax2/include/iax2.h<br>M channels/sig_analog.c<br>M channels/sig_analog.h<br>M main/asterisk.c<br>M main/bridge.c<br>M main/channel.c<br>M res/res_mutestream.c<br>M res/res_tonedetect.c<br>12 files changed, 22 insertions(+), 22 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/01/18601/1</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 ae17bca..765e9a4 100644</span><br><span>--- a/apps/app_confbridge.c</span><br><span>+++ b/apps/app_confbridge.c</span><br><span>@@ -1732,7 +1732,7 @@</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 3cf2343..2357c99 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/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/&lt;channel#&gt;[c|r&lt;cadance#&gt;|d][/extension])</literal></para></span><br><span style="color: hsl(0, 100%, 40%);">-         <para><literal>Dial(DAHDI/(g|G|r|R)&lt;group#(0-63)&gt;[c|r&lt;cadance#&gt;|d][/extension])</literal></para></span><br><span style="color: hsl(120, 100%, 40%);">+          <para><literal>Dial(DAHDI/&lt;channel#&gt;[c|r&lt;cadence#&gt;|d][/extension])</literal></para></span><br><span style="color: hsl(120, 100%, 40%);">+               <para><literal>Dial(DAHDI/(g|G|r|R)&lt;group#(0-63)&gt;[c|r&lt;cadence#&gt;|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/chan_iax2.c b/channels/chan_iax2.c</span><br><span>index 6d76dc5..8e97ef1 100644</span><br><span>--- a/channels/chan_iax2.c</span><br><span>+++ b/channels/chan_iax2.c</span><br><span>@@ -14310,7 +14310,7 @@</span><br><span>               close(com[1]);</span><br><span>               close(com[0]);</span><br><span>               if (doabort) {</span><br><span style="color: hsl(0, 100%, 40%);">-                  /* Don't interpret anything, just abort.  Not sure what th epoint</span><br><span style="color: hsl(120, 100%, 40%);">+                 /* Don't interpret anything, just abort.  Not sure what the point</span><br><span>                          of undeferring dtmf on a hung up channel is but hey whatever */</span><br><span>                    if (!old && chan)</span><br><span>                            ast_channel_undefer_dtmf(chan);</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 ea507fe..842b450 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/main/asterisk.c b/main/asterisk.c</span><br><span>index b965a4d..45f3237 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>@@ -3554,7 +3554,7 @@</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>@@ -4063,7 +4063,7 @@</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/res/res_mutestream.c b/res/res_mutestream.c</span><br><span>index 8040a3a..df1e148 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><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: 1 </div>
<div style="display:none"> Gerrit-Owner: N A <mail@interlinked.x10host.com> </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>