[asterisk-commits] branch mattf/asterisk-ss7 r37883 - /team/mattf/asterisk-ss7/channels/chan_zap.c

[asterisk-commits at lists.digium.com]

Author: mattf
Date: Tue Jul 18 11:47:17 2006
New Revision: 37883

URL: http://svn.digium.com/view/asterisk?rev=37883&view=rev
Log:
More chan_zap/libss7 integration

Modified:
    team/mattf/asterisk-ss7/channels/chan_zap.c

Modified: team/mattf/asterisk-ss7/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/mattf/asterisk-ss7/channels/chan_zap.c?rev=37883&r1=37882&r2=37883&view=diff
==============================================================================
--- team/mattf/asterisk-ss7/channels/chan_zap.c (original)
+++ team/mattf/asterisk-ss7/channels/chan_zap.c Tue Jul 18 11:47:17 2006
@@ -2631,7 +2631,7 @@
 			}
 		}
 #endif
-		if (p->sig && (p->sig != SIG_PRI))
+		if (p->sig && ((p->sig != SIG_PRI) && (p->sig != SIG_SS7)))
 			res = zt_set_hook(p->subs[SUB_REAL].zfd, ZT_ONHOOK);
 		if (res < 0) {
 			ast_log(LOG_WARNING, "Unable to hangup line %s\n", ast->name);
@@ -2682,7 +2682,7 @@
 		update_conf(p);
 		reset_conf(p);
 		/* Restore data mode */
-		if (p->sig == SIG_PRI) {
+		if ((p->sig == SIG_PRI) || (p->sig == SIG_SS7)) {
 			x = 0;
 			ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
 		}
@@ -5240,7 +5240,7 @@
 				i->dsp_features = features & ~DSP_PROGRESS_TALK;
 #ifdef HAVE_PRI
 				/* We cannot do progress detection until receives PROGRESS message */
-				if (i->outgoing && (i->sig == SIG_PRI)) {
+				if (i->outgoing && ((i->sig == SIG_PRI) || (i->sig == SIG_SS7))) {
 					/* Remember requested DSP features, don't treat
 					   talking as ANSWER */
 					features = 0;
@@ -6659,6 +6659,7 @@
 			zt_set_hook(i->subs[SUB_REAL].zfd, ZT_ONHOOK);
 			break;
 		case SIG_PRI:
+		case SIG_SS7:
 			zt_disable_ec(i);
 			res = tone_zone_play_tone(i->subs[SUB_REAL].zfd, -1);
 			break;