Run setstate after setting all appropriate variables including callerid

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_h323.c

@@ -656,7 +656,6 @@ static struct ast_channel *oh323_new(struct oh323_pvt *i, int state, const char
 		fmt = ast_best_codec(ch->nativeformats);
 		ch->type = type;
 		ch->fds[0] = ast_rtp_fd(i->rtp);
-		ast_setstate(ch, state);
 		
 		if (state == AST_STATE_RING)
 			ch->rings = 1;
@@ -700,6 +699,7 @@ static struct ast_channel *oh323_new(struct oh323_pvt *i, int state, const char
 			strncpy(ch->accountcode, i->accountcode, sizeof(ch->accountcode)-1);
 		if (i->amaflags)
 			ch->amaflags = i->amaflags;
+		ast_setstate(ch, state);
 		if (state != AST_STATE_DOWN) {
 			if (ast_pbx_start(ch)) {
 				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ch->name);

channels/chan_sip.c

@@ -1557,7 +1557,6 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 			tmp->fds[2] = ast_rtp_fd(i->vrtp);
 			tmp->fds[3] = ast_rtcp_fd(i->vrtp);
 		}
-		ast_setstate(tmp, state);
 		if (state == AST_STATE_RING)
 			tmp->rings = 1;
 		tmp->adsicpe = AST_ADSI_UNAVAILABLE;
@@ -1605,7 +1604,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
 		if (strlen(i->domain)) {
 			pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
 		}
-						
+		ast_setstate(tmp, state);
 		if (state != AST_STATE_DOWN) {
 			if (ast_pbx_start(tmp)) {
 				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);