Fix various typos, reported by Lintian

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index d99c00d27d036ac7383892bb166f4ad3f62668f8..4fd805fc7e8829a88c82ffa05f96e0efc9bea0e9 100644 (file)
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -3544,7 +3544,7 @@ plain_message:
                                                ast_log(LOG_DEBUG, "VOLGAIN: Stored at: %s.%s - Level: %.4f - Mailbox: %s\n", attach, format, vmu->volgain, mailbox);
                                        }
                                } else {
-                                       ast_log(LOG_WARNING, "Sox failed to reencode %s.%s: %s (have you installed support for all sox file formats?)\n", attach, format,
+                                       ast_log(LOG_WARNING, "Sox failed to re-encode %s.%s: %s (have you installed support for all sox file formats?)\n", attach, format,
                                                soxstatus == 1 ? "Problem with command line options" : "An error occurred during file processing");
                                        ast_log(LOG_WARNING, "Voicemail attachment will have no volume gain.\n");
                                }

diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 9eef9e3c199154895d6dd6583793425f643303b5..e9cf2db4d512eb4b7202b3dbd0a228cb371a3f3e 100644 (file)
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -3416,10 +3416,10 @@ static int dahdi_setoption(struct ast_channel *chan, int option, void *data, int
        case AST_OPTION_ECHOCAN:
                cp = (char *) data;
                if (*cp) {
-                       ast_log(LOG_DEBUG, "Enabling echo cancelation on %s\n", chan->name);
+                       ast_log(LOG_DEBUG, "Enabling echo cancellation on %s\n", chan->name);
                        dahdi_enable_ec(p);
                } else {
-                       ast_log(LOG_DEBUG, "Disabling echo cancelation on %s\n", chan->name);
+                       ast_log(LOG_DEBUG, "Disabling echo cancellation on %s\n", chan->name);
                        dahdi_disable_ec(p);
                }
                break;
@@ -5167,7 +5167,7 @@ static struct ast_frame  *dahdi_read(struct ast_channel *ast)
        
        /* Hang up if we don't really exist */
        if (index < 0)  {
-               ast_log(LOG_WARNING, "We dont exist?\n");
+               ast_log(LOG_WARNING, "We don't exist?\n");
                ast_mutex_unlock(&p->lock);
                return NULL;
        }

diff --git a/channels/chan_misdn.c b/channels/chan_misdn.c
index 9d60ac38414734c3d5372697aa920e9894b70a40..95aff770ab0c323383cb98a66157ed35a2fa41af 100644 (file)
--- a/channels/chan_misdn.c
+++ b/channels/chan_misdn.c
@@ -5276,9 +5276,9 @@ static int load_module(void)
                "    a - Have Asterisk detect DTMF tones on called channel\n"
                "    c - Make crypted outgoing call, optarg is keyindex\n"
                "    d - Send display text to called phone, text is the optarg\n"
-               "    e - Perform echo cancelation on this channel,\n"
+               "    e - Perform echo cancellation on this channel,\n"
                "        takes taps as optarg (32,64,128,256)\n"
-               "   e! - Disable echo cancelation on this channel\n"
+               "   e! - Disable echo cancellation on this channel\n"
                "    f - Enable fax detection\n"
                "    h - Make digital outgoing call\n"
                "   h1 - Make HDLC mode digital outgoing call\n"

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4e8005e0e3a6a39277b25a9faaa7e7549711d56c..f2585a487e715eb16dffe624074e4e69c73ca434 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1212,7 +1212,7 @@ struct sip_registry {
        struct sip_pvt *call;           /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
        enum sipregistrystate regstate; /*!< Registration state (see above) */
        unsigned int needdns:1; /*!< Set if we need a new dns lookup before we try to transmit */
-       time_t regtime;         /*!< Last succesful registration time */
+       time_t regtime;         /*!< Last successful registration time */
        int callid_valid;               /*!< 0 means we haven't chosen callid for this registry yet. */
        unsigned int ocseq;             /*!< Sequence number we got to for REGISTERs for this registry */
        struct sockaddr_in us;          /*!< Who the server thinks we are */
@@ -9117,7 +9117,7 @@ static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *
                return AUTH_CHALLENGE_SENT;
        } 
        if (good_response) {
-               append_history(p, "AuthOK", "Auth challenge succesful for %s", username);
+               append_history(p, "AuthOK", "Auth challenge successful for %s", username);
                return AUTH_SUCCESSFUL;
        }
 
@@ -13753,7 +13753,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                                        }
                                }
                        } else
-                               ast_log(LOG_NOTICE, "Dont know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_inet_ntoa(p->sa.sin_addr));
+                               ast_log(LOG_NOTICE, "Don't know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_inet_ntoa(p->sa.sin_addr));
                }
        } else {        
                /* Responses to OUTGOING SIP requests on INCOMING calls 
@@ -16463,7 +16463,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
                }
                if (!p->initreq.headers) {
                        if (option_debug)
-                               ast_log(LOG_DEBUG, "That's odd...  Got a response on a call we dont know about. Cseq %d Cmd %s\n", seqno, cmd);
+                               ast_log(LOG_DEBUG, "That's odd...  Got a response on a call we don't know about. Cseq %d Cmd %s\n", seqno, cmd);
                        ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
                        return 0;
                }

diff --git a/res/res_agi.c b/res/res_agi.c
index 000575762320f6f673e39c4202a8618c49e8c5f8..27d363429d6dd452cdd20e2d33cc2670b7dca372 100644 (file)
--- a/res/res_agi.c
+++ b/res/res_agi.c
@@ -1651,7 +1651,7 @@ static char usage_recordfile[] =
 " to the offset without exceeding the end of the file.  \"silence\" is the number\n"
 " of seconds of silence allowed before the function returns despite the\n"
 " lack of dtmf digits or reaching timeout.  Silence value must be\n"
-" preceeded by \"s=\" and is also optional.\n";
+" preceded by \"s=\" and is also optional.\n";
 
 static char usage_autohangup[] =
 " Usage: SET AUTOHANGUP <time>\n"

diff --git a/res/res_jabber.c b/res/res_jabber.c
index d1729f17936c00ada1ef55f3b537f91d91e83dac..258a98352cbeedd6dad9c89962d7486f4d3babff 100644 (file)
--- a/res/res_jabber.c
+++ b/res/res_jabber.c
@@ -676,16 +676,16 @@ static int aji_act_hook(void *data, int type, iks *node)
        case IKS_PAK_S10N:
                aji_handle_subscribe(client, pak);
                if (option_debug)
-                       ast_log(LOG_DEBUG, "JABBER: I Dont know S10N subscribe!!\n");
+                       ast_log(LOG_DEBUG, "JABBER: I Don't know S10N subscribe!!\n");
                break;
        case IKS_PAK_IQ:
                if (option_debug)
-                       ast_log(LOG_DEBUG, "JABBER: I Dont have an IQ!!!\n");
+                       ast_log(LOG_DEBUG, "JABBER: I Don't have an IQ!!!\n");
                aji_handle_iq(client, node);
                break;
        default:
                if (option_debug)
-                       ast_log(LOG_DEBUG, "JABBER: I Dont know %i\n", pak->type);
+                       ast_log(LOG_DEBUG, "JABBER: I Don't know %i\n", pak->type);
                break;
        }