Merged revisions 175952 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines
  
  Merged revisions 175921 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
    
    fix mis-spelling of the word registered.
    Reported by De_Mon on #asterisk-dev.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -411,7 +411,7 @@ enum check_auth_result {
 
 /*! \brief States for outbound registrations (with register= lines in sip.conf */
 enum sipregistrystate {
-	REG_STATE_UNREGISTERED = 0,	/*!< We are not registred 
+	REG_STATE_UNREGISTERED = 0,	/*!< We are not registered 
 		 *  \note Initial state. We should have a timeout scheduled for the initial
 		 * (or next) registration transmission, calling sip_reregister
 		 */
@@ -21013,7 +21013,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
 	*/
 	if (create_addr(p, host, NULL, 1)) {
 		*cause = AST_CAUSE_UNREGISTERED;
-		ast_debug(3, "Cant create SIP call - target device not registred\n");
+		ast_debug(3, "Cant create SIP call - target device not registered\n");
 		dialog_unlink_all(p, TRUE, TRUE);
 		dialog_unref(p, "unref dialog p UNREGISTERED");
 		/* sip_destroy(p); */

channels/chan_unistim.c

@@ -1117,7 +1117,7 @@ static void close_client(struct unistimsession *s)
 		cur = cur->next;
 	}
 	if (cur) {				      /* Session found ? */
-		if (cur->device) {	      /* This session was registred ? */
+		if (cur->device) {	      /* This session was registered ? */
 			s->state = STATE_CLEANING;
 			if (unistimdebug)
 				ast_verb(0, "close_client session %p device %p lines %p sub %p\n",
@@ -3324,7 +3324,7 @@ static void init_phone_step2(struct unistimsession *pte)
 			for (i = 1; i < 6; i++)
 				send_favorite(i, 0, pte, "");
 			send_text(TEXT_LINE0, TEXT_NORMAL, pte, "Sorry, this phone is not");
-			send_text(TEXT_LINE1, TEXT_NORMAL, pte, "registred in unistim.cfg");
+			send_text(TEXT_LINE1, TEXT_NORMAL, pte, "registered in unistim.cfg");
 			strcpy(tmp, "MAC = ");
 			strcat(tmp, pte->macaddr);
 			send_text(TEXT_LINE2, TEXT_NORMAL, pte, tmp);
@@ -3419,7 +3419,7 @@ static void process_request(int size, unsigned char *buf, struct unistimsession
 	if (memcmp(buf + SIZE_HEADER, packet_recv_pick_up, sizeof(packet_recv_pick_up)) == 0) {
 		if (unistimdebug)
 			ast_verb(0, "Handset off hook\n");
-		if (!pte->device)	       /* We are not yet registred (asking for a TN in AUTOPROVISIONING_TN) */
+		if (!pte->device)	       /* We are not yet registered (asking for a TN in AUTOPROVISIONING_TN) */
 			return;
 		pte->device->receiver_state = STATE_OFFHOOK;
 		if (pte->device->output == OUTPUT_HEADPHONE)

doc/unistim.txt

@@ -64,7 +64,7 @@ Autoprovisioning :
 - This feature must only be used on a trusted network. It's very insecure : all unistim phones
   will be able to use your asterisk pbx.
 - You must add an entry called [template]. Each new phones will be based on this profile.
-- You must set a least line=>. This value will be incremented when a new phone is registred.
+- You must set a least line=>. This value will be incremented when a new phone is registered.
   device= must not be specified. By default, the phone will asks for a number. It will be added into
   the dialplan. Add extension=line for using the generated line number instead.
  Example :

include/asterisk/manager.h

@@ -147,8 +147,8 @@ int ast_manager_register2(
 	const char *synopsis,
 	const char *description);
 
-/*! \brief Unregister a registred manager command 
-	\param action Name of registred Action:
+/*! \brief Unregister a registered manager command 
+	\param action Name of registered Action:
 */
 int ast_manager_unregister( char *action );