Fixed more stuff for clearchannel mode in app_dial

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@678 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 25 insertions, 10 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index e68f5137e..472f14436 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -63,6 +63,7 @@ static char *descrip =
 "      'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
 "      'm' -- provide hold music to the calling party until answered.\n"
 "      'd' -- data-quality (modem) call (minimum delay).\n"
+"      'c' -- clear-channel data call (PRI-PRI only).\n"
 "      'H' -- allow caller to hang up by hitting *.\n"
 "      'C' -- reset call detail record for this call.\n"
 "      'P[(x)]' -- privacy mode, using 'x' as database if provided.\n"
@@ -82,6 +83,7 @@ struct localuser {
 	int ringbackonly;
 	int musiconhold;
 	int dataquality;
+	int clearchannel;
 	int allowdisconnect;
 	struct localuser *next;
 };
@@ -427,6 +429,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			if (strchr(transfer, 'H'))
 				tmp->allowdisconnect = 1;
                         else    tmp->allowdisconnect = 0;
+			if (strchr(transfer, 'c'))
+				tmp->clearchannel = 1;
+                        else    tmp->clearchannel = 0;
 		}
 		strncpy(numsubst, number, sizeof(numsubst)-1);
 		/* If we're dialing by extension, look at the extension to know what to dial */
@@ -543,18 +548,14 @@ static int dial_exec(struct ast_channel *chan, void *data)
 		if (!strcmp(chan->type,"Zap"))
 		{
 			int x = 2;
-			if (tmp->dataquality) x = 0;
+			if (tmp->dataquality | tmp->clearchannel) x = 0;
 			ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
-			x = 0;
-			ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
 		}			
 		if (!strcmp(peer->type,"Zap"))
 		{
 			int x = 2;
 			if (tmp->dataquality) x = 0;
 			ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
-			x = 0;
-			ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
 		}			
 		hanguptree(outgoing, peer);
 		outgoing = NULL;
@@ -577,7 +578,19 @@ static int dial_exec(struct ast_channel *chan, void *data)
  			ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
  			ast_channel_sendurl( peer, url );
  		} /* /JDG */
-		res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->dataquality);
+		if (tmp->clearchannel)
+		{
+			int x = 0;
+			ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
+			ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
+		}
+		res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->clearchannel);
+		if (tmp->clearchannel)
+		{
+			int x = 1;
+			ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
+			ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
+		}
 		ast_hangup(peer);
 	}	
 out:
diff --git a/channels/chan_zap.c b/channels/chan_zap.c
index a6a49f3bd..360905787 100755
--- a/channels/chan_zap.c
+++ b/channels/chan_zap.c
@@ -1433,6 +1433,9 @@ static int zt_hangup(struct ast_channel *ast)
 	
 	index = zt_get_index(ast, p, 1);
 
+	x = 1;
+	ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
+
 	restore_gains(p);
 	
 	if (p->dsp)
@@ -1617,8 +1620,6 @@ static int zt_hangup(struct ast_channel *ast)
 		x = 0;
 		ast_channel_setoption(ast,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
 		ast_channel_setoption(ast,AST_OPTION_TDD,&x,sizeof(char),0);
-		x = 1;
-		ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
 		p->didtdd = 0;
 		p->cidspill = NULL;
 		p->callwaitcas = 0;
@@ -1864,6 +1865,7 @@ int	x;
 		{		
 			ast_log(LOG_DEBUG, "Set option AUDIO MODE, value: OFF(0) on %s\n",chan->name);
 			x = 0;
+			zt_disable_ec(p);
 		}
 		else
 		{		
@@ -5461,10 +5463,10 @@ static void *pri_dchannel(void *vpri)
 						ast_verbose(VERBOSE_PREFIX_2 "Restart on requested on entire span %d\n", pri->span);
 					for (x=1;x <= pri->channels;x++)
 						if ((x != pri->dchannel) && pri->pvt[x]) {
-							ast_pthread_mutex_lock(&pri->pvt[chan]->lock);
+							ast_pthread_mutex_lock(&pri->pvt[x]->lock);
  							if (pri->pvt[x]->owner)
 								pri->pvt[x]->owner->_softhangup |= AST_SOFTHANGUP_DEV;
-							ast_pthread_mutex_unlock(&pri->pvt[chan]->lock);
+							ast_pthread_mutex_unlock(&pri->pvt[x]->lock);
 						}
 				}
 				break;