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Merged revisions 175921 via svnmerge from

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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/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Michiel van Baak
2009-02-16 00:26:59 +00:00
parent 8c75380f52
commit 115c6abef4
4 changed files with 8 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files
channels/chan_sip.c
+2 -2
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@@ -694,7 +694,7 @@ 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
*/
@@ -21967,7 +21967,7 @@ 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
+3 -3
View File
@@ -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
+1 -1
View File
@@ -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
+2 -2
View File
@@ -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 );