[asterisk-commits] tzafrir: branch 1.4 r273640 - in /branches/1.4: apps/ channels/ res/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jul 2 10:54:21 CDT 2010


Author: tzafrir
Date: Fri Jul  2 10:54:17 2010
New Revision: 273640

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=273640
Log:
Fix various typos, reported by Lintian

Modified:
    branches/1.4/apps/app_voicemail.c
    branches/1.4/channels/chan_dahdi.c
    branches/1.4/channels/chan_misdn.c
    branches/1.4/channels/chan_sip.c
    branches/1.4/res/res_agi.c
    branches/1.4/res/res_jabber.c

Modified: branches/1.4/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/apps/app_voicemail.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/apps/app_voicemail.c (original)
+++ branches/1.4/apps/app_voicemail.c Fri Jul  2 10:54:17 2010
@@ -3544,7 +3544,7 @@
 						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");
 				}

Modified: branches/1.4/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_dahdi.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/channels/chan_dahdi.c (original)
+++ branches/1.4/channels/chan_dahdi.c Fri Jul  2 10:54:17 2010
@@ -3416,10 +3416,10 @@
 	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 @@
 	
 	/* 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;
 	}

Modified: branches/1.4/channels/chan_misdn.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_misdn.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/channels/chan_misdn.c (original)
+++ branches/1.4/channels/chan_misdn.c Fri Jul  2 10:54:17 2010
@@ -5276,9 +5276,9 @@
 		"    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"

Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Jul  2 10:54:17 2010
@@ -1212,7 +1212,7 @@
 	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 @@
 		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 @@
 					}
 				}
 			} 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 @@
 		}
 		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;
 		}

Modified: branches/1.4/res/res_agi.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/res/res_agi.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/res/res_agi.c (original)
+++ branches/1.4/res/res_agi.c Fri Jul  2 10:54:17 2010
@@ -1651,7 +1651,7 @@
 " 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"

Modified: branches/1.4/res/res_jabber.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/res/res_jabber.c?view=diff&rev=273640&r1=273639&r2=273640
==============================================================================
--- branches/1.4/res/res_jabber.c (original)
+++ branches/1.4/res/res_jabber.c Fri Jul  2 10:54:17 2010
@@ -676,16 +676,16 @@
 	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;
 	}
 	




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