Index: channels/chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.677
diff -u -r1.677 chan_sip.c
--- channels/chan_sip.c	17 Mar 2005 23:03:07 -0000	1.677
+++ channels/chan_sip.c	18 Mar 2005 13:53:18 -0000
@@ -2078,7 +2078,6 @@
 		res = sip_sipredirect(p, dest);
 	else
 		res = transmit_refer(p, dest);
-	res = transmit_refer(p, dest);
 	ast_mutex_unlock(&p->lock);
 	return res;
 }
@@ -10232,16 +10231,16 @@
 
 #define DEFAULT_MAX_FORWARDS	70
 
-/* This is 302 sipredirect function coded by Martin Pycko (m78pl@yahoo.com) */
+/*--- sip_sipredirect: Transfer call before connect with a 302 redirect ---*/
+/* Called by the transfer() dialplan application through the sip_transfer() */
+/* pbx interface function if the call is in ringing state */
+/* coded by Martin Pycko (m78pl@yahoo.com) */
 static int sip_sipredirect(struct sip_pvt *p, char *dest)
 {
 	char *cdest;
 	char *extension, *host, *port;
 	char tmp[80];
-	if (!dest || ast_strlen_zero(dest)) {
-		ast_log(LOG_WARNING, "This application requires these arguments: SIPRedirect(extension[@host[:port]])\n");
-		return 0;
-	}
+	
 	cdest = ast_strdupa(dest);
 	if (!cdest) {
 		ast_log(LOG_ERROR, "Problem allocating the memory\n");
@@ -10268,7 +10267,7 @@
 			memset(lhost, 0, sizeof(lhost));
 			memset(lport, 0, sizeof(lport));
 			localtmp++;
-			/* This is okay becuase lhost and lport are as big as tmp */
+			/* This is okey because lhost and lport are as big as tmp */
 			sscanf(localtmp, "%[^<>:; ]:%[^<>:; ]", lhost, lport);
 			if (!strlen(lhost)) {
 				ast_log(LOG_ERROR, "Can't find the host address\n");
@@ -10299,7 +10298,7 @@
 		p->maxforwards = DEFAULT_MAX_FORWARDS - 1;
 	}
 	if (p->maxforwards > -1) {
-		snprintf(p->our_contact, sizeof(p->our_contact), "redirect <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
+		snprintf(p->our_contact, sizeof(p->our_contact), "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
 		transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 1);
 	} else {
 		transmit_response(p, "483 Too Many Hops", &p->initreq);
Index: apps/app_transfer.c
===================================================================
RCS file: /usr/cvsroot/asterisk/apps/app_transfer.c,v
retrieving revision 1.7
diff -u -r1.7 app_transfer.c
--- apps/app_transfer.c	19 Feb 2005 17:34:06 -0000	1.7
+++ apps/app_transfer.c	18 Mar 2005 13:53:18 -0000
@@ -29,10 +29,14 @@
 static char *synopsis = "Transfer caller to remote extension";
 
 static char *descrip = 
-"  Transfer([Tech/]dest):  Requests the remote caller be transferred\n"
-"a given extension. Returns -1 on hangup, or 0 on completion\n"
-"regardless of whether the transfer was successful.  If the transfer\n"
-"was *not* supported or successful and there exists a priority n + 101,\n"
+"  Transfer([Tech/]dest):  Requests the remote caller be transfered\n"
+"to a given extension. If TECH (SIP, IAX2, LOCAL etc) is used, only\n"
+"an incoming call with the same channel technology will be transfered.\n"
+"Note that for SIP, if you transfer before call is setup, a 302 redirect\n"
+"SIP message will be returned to the caller.\n"
+"Returns -1 on hangup, or 0 on completion regardless of whether the\n"
+"transfer was successful.  If the transfer was *not* supported or\n"
+"successful and there exists a priority n + 101,\n"
 "then that priority will be taken next.\n" ;
 
 STANDARD_LOCAL_USER;